-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.34 ===
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- Fix typo in the docs.
-
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
- If no stream was found before receiving EOS, post an error message.
- Fixes #561924.
-
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-05-14 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c: (gst_theora_enc_init),
- (theora_buffer_from_packet), (theora_push_packet),
- (theora_enc_sink_event), (theora_enc_is_discontinuous),
- (theora_enc_chain):
- Parse segment events.
- Pass incomming buffer timestamps to outgoing buffers.
- Use the running_time to construct the granulepos.
- Fixes #562163.
+ * configure.ac:
+ releasing 0.10.34, "Lemmings"
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.33 ===
- * gst/playback/gstplaybin2.c: (activate_group):
- Fix buffer-duration property.
+2011-05-10 09:32:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.33
+ Highlights:
+ - support for 16-bit-per-component video formats
+ - playbin2 fixes and improvements for custom and non-raw sinks
+ - oggmux muxes based on running time now
+ - many other fixes and improvements
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
- (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- Really fix audiosink drain handling by keeping track of the running_time
- of the last sample.
+2011-04-30 17:35:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 Michael Smith <msmith@songbirdnest.com>
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * po/da.po:
+ * po/de.po:
+ * po/fr.po:
+ * po/uk.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.32.4 pre-release
- * gst/playback/gstplaybin2.c:
- Add notification of current stream. Add ability to configure buffer
- sizes.
- * gst/playback/gsturidecodebin.c:
- Add ability to configure buffer sizes for streaming mode.
- Bug #561734.
+2011-04-30 17:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ gst: update orc-generated disted C backup code to orc 0.4.14
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- Time is already in running_time. Remove base_time handling. Fixes
- audiosinks not draining and thus chopping some audio in the end.
+2011-04-27 12:09:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 David Schleef <ds@schleef.org>
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * po/bg.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sl.po:
+ * po/tr.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.32.3 pre-release
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- If we're muxing a dirac stream, flush the page after every picture.
+2011-04-24 18:45:40 -0700 David Schleef <ds@schleef.org>
-2008-11-24 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/vs_image.c:
+ videoscale: Fix off-by-one error in previous commit
+ Fix for 7c0b702e. It helps to get your j+1's right.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- Add one log message to check for audio_drained. Sync one log message
- with the condition. Send EOS after draining audio in pull mode.
+2011-04-24 18:16:20 -0700 David Schleef <ds@schleef.org>
-2008-11-24 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
- Use gst_buffer_try_new_and_alloc() and fail properly if the
- allocation failed. This prevents abort() if downstream elements
- request an insane amount of memory.
-
-2008-11-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/volume/gstvolume.c: (volume_choose_func),
- (volume_update_volume), (gst_volume_set_volume),
- (gst_volume_get_volume), (gst_volume_set_mute),
- (gst_volume_class_init), (gst_volume_init),
- (volume_process_double), (volume_process_float),
- (volume_process_int32), (volume_process_int32_clamp),
- (volume_process_int24), (volume_process_int24_clamp),
- (volume_process_int16), (volume_process_int16_clamp),
- (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
- (volume_transform_ip), (volume_set_property),
- (volume_get_property):
- * gst/volume/gstvolume.h:
- Cleanup volume, define and use default values.
- Recalculate new volume and mute setup before processing. Fixes #561789.
+ * gst/videoscale/vs_image.c:
+ videoscale: Fix ARGB bilinear scaling
+ Fixes #648548. Orc generates bad code for
+ gst_videoscale_orc_resample_merge_bilinear_u32, so we'll use the
+ slightly slower two-stage process. I'd fix Orc, but it's hard to
+ get excited about fixing a feature that I'm planning to deprecate
+ and replace.
- * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
- Add controller unit test. Patch by: Jonathan Matthew
- Fix bogus test that messed with basetransform's internal state.
+2011-04-23 13:42:23 -0700 David Schleef <ds@schleef.org>
-2008-11-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/vs_image.c:
+ videoscale: hack to fix invalid reads in linear
+ https://bugzilla.gnome.org/show_bug.cgi?id=633837
- * gst/videorate/gstvideorate.c:
- Add jpeg and png image media types to the caps. Fixes #561436.
+2011-04-23 12:46:09 -0700 David Schleef <ds@schleef.org>
-2008-11-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: protect 4tap from out-of-bounds reads
+ https://bugzilla.gnome.org/show_bug.cgi?id=633837
- * gst/playback/gstplaysink.c: (gen_audio_chain):
- Don't post an error when we can't configure the volume but post a
- warning instead. Fixes #561780.
+2011-04-24 14:03:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-21 David Schleef <ds@schleef.org>
+ * common:
+ Automatic update of common submodule
+ From c3cafe1 to 46dfcea
- Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
+2011-04-23 12:44:50 -0700 David Schleef <ds@schleef.org>
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add a zone plate pattern generator based on BBC R&D Report
- 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
- kx2=20 ky2=20 kt=1'.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: use simpler scaling method for small images
+ https://bugzilla.gnome.org/show_bug.cgi?id=633837
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2011-04-14 09:32:19 +0200 Marc Plano-Lesay <marc.planolesay@gmail.com>
- * gst/playback/gstdecodebin2.c:
- Fix random fat-fingering making this not compile.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: fix unused-but-set-variable warnings with gcc 4.6
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2011-04-22 13:55:20 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
- * gst/playback/gstdecodebin2.c:
- If the top-level type of the stream is plain text, don't try to decode
- it, matching behaviour of decodebin.
- * gst/playback/gstplaysink.c:
- If we fail to generate a text chain (e.g. due to missing optional
- plugins), don't crash.
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ tag: exif: register common tags from tag library
+ Exif uses tags like image-vertical-ppi or image-horizontal-ppi which are
+ registered in gst_tag_register_musicbrainz_tags(), but neither GstExifReader
+ nor GstExifWriter register them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=648459
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2011-04-24 12:16:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Fix win32 build. Oops.
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: update some FIXMEs for 0.11
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2011-04-21 14:11:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Use WSAGetLastError() rather than errno/h_errno on win32.
+ * tests/check/elements/videoscale.c:
+ tests: add unit test for basetransform/videoscale negotiation regression
+ Turn Rene's test pipeline into a unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=648220
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-11-25 17:01:53 +0100 Håvard Graff <havard.graff@.eu.tandberg.int>
- * gst-libs/gst/riff/riff-media.c:
- Support WMA Lossless properly.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: make sure to not start if the may_start flag is FALSE
+ Fixes #635784
-2008-11-18 David Schleef <ds@schleef.org>
+2011-04-18 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add "colorspec" property, specifying whether to generate BT.601
- or BT.709 video. This only affects YCbCr values, not RGB, since
- if you're generating a 709 test pattern, presumably you want
- 709 RGB primaries, not 601. Also add "smpte75" pattern, which
- uses 75% colors instead of 100%, since this is often more useful
- for testing (and also follows the SMPTE EG-1 guideline).
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: If NULL caps are passed to buffer_alloc() do fallback allocation
+ Fixes bug #647857.
-2008-11-18 Alessandro Decina <alessandro.d@gmail.com>
+2011-04-18 10:19:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c:
- Add a "sink-caps" property to decodebin like it's done for decodebin2.
- Fixes #560380.
+ * tests/check/pipelines/oggmux.c:
+ oggmux: Remove bus GSource to prevent a valgrind warning
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2011-04-18 09:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c:
- Guard against a NULL dereference I somehow encountered -
- with a FLUSH_STOP arriving either before basetransform _start(),
- or after _stop().
+ * tests/check/pipelines/gio.c:
+ gio: Remove the bus GSource from the main context
+ Prevents a valgrind warning about possibly leaked memory,
+ see bug #647763.
- * gst/typefind/gsttypefindfunctions.c:
- Make sure we never jump backwards when typefinding corrupt mov files.
+2011-04-17 19:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: remove gst_init() for g-i scanner here again as well to avoid problems with -Wl,--as-needed
- * gst-libs/gst/interfaces/propertyprobe.c:
- Fix random type causing a docs warning.
+2011-04-17 17:59:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-14 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/fft/Makefile.am:
+ fft: remove gst_init() for g-i scanner again
+ libgstfft doesn't actually use any symbols from libgstreamer, so when
+ compiling with -Wl,--as-needed it won't even link to it, which can
+ cause failures with older versions of g-i that ignore the --pkg
+ arguments.
+ Should fix PPA build failure on Ubuntu Maverick
- * sys/v4l/gstv4l.c:
- Give it a minimal rank for autovideosrc.
+2011-04-16 16:31:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-13 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Always hold the class-global pango mutex when using pango API
- * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
- (plugin_init):
- Improve typefinding of ISO JPEG2000 mime types.
+2011-04-16 16:23:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttimeoverlay.c:
+ {time,clock}overlay: Hold the class-global pango mutex when changing the pango context
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.h:
- Avoid typechecking when we do trivial casts.
- Move error handling out of the main program flow.
- Sneak in the display-region caps property, not completely correct yet.
- Cache the width/height in buffer_alloc instead of parsing it from the
- caps all the time.
-
-2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (deactivate_group):
- don't try to unlink the selector sinkpad when we don't have it yet. This
- can happen if an error occured before the group was complete.
-
-2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
- (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
- (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
- (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
- (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
- (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
- (gst_rtp_buffer_get_extension_data),
- (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
- (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
- (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
- (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
- (gst_rtp_buffer_get_payload_type),
- (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
- (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
- (gst_rtp_buffer_set_timestamp),
- (gst_rtp_buffer_get_payload_subbuffer),
- (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
- Avoid expensive type checks we already did as part of the
- _validate() function that should be called first.
-
-2008-11-11 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
- (gst_base_rtp_depayload_push_full),
- (gst_base_rtp_depayload_set_gst_timestamp):
- Fix some cases where a newsegment event was not sent.
-
-2008-11-11 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (activate_group):
- Catch state change errors and stop from the uridecodebin elements
- instead of trying to continue in vain.
-
-2008-11-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-16 16:21:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
- (gst_base_audio_sink_callback):
- Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
- for the latency to expire, fixes #559567.
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttimeoverlay.c:
+ {clock,time}overlay: Only set the global pango context options once in class_init
+ Instead of doing it over and over again when instantiating a new instance.
-2008-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2011-04-16 16:18:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c:
- Change author string after seeing output of gst-inspector.
+ * ext/pango/gsttextoverlay.c:
+ pango: Create a new pango context for every subclass
+ timeoverlay/clockoverlay are setting some global options
+ on the context that shouldn't be used for the generic textoverlay.
-2008-11-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-16 16:03:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
- Don't try to do crazy things when we only have a text pad without a
- video pad. Fixes #559478.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.32.2 pre-release
-2008-11-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-16 15:58:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Fix case where we don't have a range for the rates or channels as is the
- case with truespeech.
+ * gst/adder/gstadderorc-dist.c:
+ * gst/adder/gstadderorc-dist.h:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.h:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ gst: update disted orc backup code
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-16 15:50:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_update_real_volume),
- (gst_volume_set_volume), (gst_volume_get_volume),
- (gst_volume_set_mute), (gst_volume_init), (volume_setup),
- (volume_transform_ip), (volume_update_mute),
- (volume_update_volume), (volume_get_property):
- * gst/volume/gstvolume.h:
- Keep negotiated state in a separate variable.
- Protect the volume and mute properties with the object lock.
- Protect modifying the transform with the transform lock.
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ docs: update documentation
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-16 15:42:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Only convert caps to string when debug is enabled.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-31 17:56:00 +0000 Thibault Saunier <thibault.saunier@collabora.co.uk>
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (gst_theora_dec_reset), (theora_dec_src_event),
- (theora_dec_sink_event), (theora_handle_type_packet):
- Copy seqnum.
- Keep events in a pending list, like vorbisdec, instead of trying
- to construct a segment event ourselves.
-
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
- (vorbis_dec_src_event), (vorbis_dec_sink_event):
- * ext/vorbis/vorbisdec.h:
- Copy seqnum.
+ * Android.mk:
+ * configure.ac:
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/gstvorbisdeclib.h:
+ vorbis: add support for using tremolo on android
+ Tremolo is an ARM-optimised version of xiph's tremor library.
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-15 13:36:39 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
- (gst_ogg_demux_loop):
- * ext/ogg/gstoggdemux.h:
- Copy seqnums around to track playback segments and messages.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggmux: prefer headers from caps to determine stream type
+ Ogg mandates the first header packet must determine a stream's type.
+ However, some streams (such as VP8) do not include such a header
+ when muxed in other containers, and thus do not include this header
+ as a buffer, but only in caps. We thus use headers from caps when
+ available to determine a new stream's type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=647856
+
+2011-04-16 11:00:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ libs: gobject-introspection scanner doesn't need to scan or update plugin info
+ Make sure the scanner doesn't load or introspect or check any plugins,
+ (especially not outside the build directory).
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-15 21:09:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Based on patch by: Matthias Kretz <kretz at kde dot org>
+ * tests/check/Makefile.am:
+ tests: list libs/struct*h files explicitly in Makefile.am
+ Hopefully makes the gentoo buildbot happy again.
- * ext/alsa/gstalsasink.c: (gst_alsasink_open),
- (gst_alsasink_prepare), (gst_alsasink_unprepare),
- (gst_alsasink_write):
- Make all access non-blocking so that we can better handle unplugging
- of usb devices. Fixes #559111
+2011-04-15 11:11:52 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: avoid foregoing READY_TO_NULL when appropriate
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+2011-04-14 22:13:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_write):
- Make the next call to poll not depend on previous calls to poll with or
- without reading from the active descriptor. Fixes #544293.
+ * gst/playback/gstplaybin2.c:
+ playbin2: ensure proper PAUSED_TO_READY cleanup
+ ... since going async to PAUSED might fail, and never making it to PAUSED
+ subsequently skips going down to READY.
+ Fixes #647781.
-2008-10-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-14 12:42:20 -0700 David Schleef <ds@schleef.org>
- Patch by: Nick Haddad <nick at haddads dot net>
+ * gst-libs/gst/video/video.c:
+ Revert "video: Remove the extensive checkings from switch"
+ This reverts commit 500d14c35c656890686574e1c041fb556df17056.
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for other fourcc codes that are commonly used for
- 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
- Fixes #558553.
+2011-04-14 13:15:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-30 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/encodebin.c:
+ encodebin: Unref encoding profiles after usage in the test
- * gst/audioresample/gstaudioresample.c:
- Return the result of parent_class->event().
+2011-04-14 12:55:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * tests/check/elements/encodebin.c:
+ encodebin: Release pads after setting the state to NULL in the unit test
+ See bug #647756.
- * gst/audioresample/gstaudioresample.c:
- (gst_audioresample_class_init), (audioresample_fixate_caps):
- Fixate the rate to the nearest supported rate instead of
- the first one. Fixes bug #549510.
+2011-04-14 12:23:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/encoding/gstencodebin.c:
+ encodebin: Set all elements to NULL and remove them from the bin when removing a source group
- * win32/common/libgstaudio.def:
- Add new symbols.
+2011-04-14 00:26:34 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
-2008-10-23 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/video/video.c:
+ video: Remove the extensive checkings from switch
+ The default case handles them already
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- Attempt to make obfuscated code clearer.
+2011-04-13 23:17:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-10-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * tests/check/libs/tag.c:
+ tests: tag: Fix typo
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/floatcast/floatcast.h:
- Move float endianness conversion macros to core. Second part of
- bug ##555196.
+2011-04-13 23:17:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
+ Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION for xmp
+ library.
+ Includes unit tests.
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h:
- Don't mark as gtk-doc docs as they aren't public.
+2011-04-13 23:16:02 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
+ Adds mapping for GST_TAG_CAPTURING_EXPOSURE_COMPENSATION for exif
+ library.
+ Includes unit tests.
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- * tests/icles/Makefile.am:
- * tests/icles/test-colorkey.c:
- Allow setting colorkey if possible. Implement property probe interface
- for optional X features (autopaint-colorkey, double-buffer and
- colorkey). Fixes #554533
+2011-04-13 23:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-10-22 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: Adds GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
+ Adds a new tag for indicating the used exposure compensation
+ level in EV used when capturing an image.
+ API: GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Remove useless buffer size assignment. It already has this value.
+2011-04-14 00:24:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/examples/encoding/gstcapslist.c:
+ * tests/examples/gio/giosrc-mounting.c:
+ * tests/examples/playrec/playrec.c:
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ tests: fix unused-but-set-variable warnings with gcc 4.6
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
- (gst_audioringbuffer_stop):
- Implement a separate activate functions to start monitoring the segments
- or, in pull mode, pulling in data.
+2011-04-13 23:57:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
- (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
- (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
- (gst_base_audio_sink_activate_pull),
- (gst_base_audio_sink_async_play),
- (gst_base_audio_sink_change_state):
- Implement pad and element convert query function.
- Activate the ringbuffer.
- Use the segment last_stop value as the offset to pull.
- Use new basesink _do_preroll() method to preroll in the pulling thread.
- Take appropriate locking in the pulling thread.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: fix unused-but-set-variable warnings with gcc 4.6
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
- * gst-libs/gst/audio/gstringbuffer.h:
- Update some docs.
+2011-04-13 23:19:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: fix unused-but-set-variable warnings with gcc 4.6
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
- * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
- Improve MXF typefinding a bit by searching for a header partition
- pack instead of just a general partition pack and checking more
- bytes for valid values.
+2011-04-13 22:59:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: fix unused-but-set-variable warning with gcc 4.6
+ We don't compare the bitrates of consecutive mp3 frames on purpose
+ here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
- * tests/icles/.cvsignore:
- update ignore file.
+2011-04-13 09:10:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/icles/Makefile.am:
- * tests/icles/test-box.c: (make_pipeline), (main):
- Add another interactive command line experimentation suite for
- dynamically boxing/cropping/saling an input video.
+ * gst-libs/gst/video/video.h:
+ docs: fix typo in video format docs
-2008-10-17 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-12 12:41:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
- (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add methods to more accuratly control the pulling thread of a
- ringbuffer.
- Add format conversion helper code to the ringbuffer.
- API: GstRingBuffer:gst_ring_buffer_activate()
- API: GstRingBuffer:gst_ring_buffer_is_active()
- API: GstRingBuffer:gst_ring_buffer_convert()
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix uninitialised variable usage and element leak
+ gcc on OSX complains about ret being used uninitialized in
+ this function, and it is right. Don't leak element ref
+ when returning early because newsegment event is not in
+ TIME format.
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-12 12:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_stop):
- Signal thread startup earlier so that we can immediatly go into pull
- mode when we have to and block on preroll.
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: do check return values of fcntl() and fstat()
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-09 19:15:23 +0200 Marc Plano-Lesay <marc.planolesay@gmail.com>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_prepare_read):
- In pull mode we want the callback to prepull a buffer we can preroll on
- even when we are not yet playing.
+ * gst/playback/gstplaybasebin.c:
+ * gst/subparse/tmplayerparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/videoscale/vs_image.c:
+ fix unused-but-set-variable warnings with gcc 4.6
+ https://bugzilla.gnome.org/show_bug.cgi?id=647294
-2008-10-16 Stefan Kost <ensonic@users.sf.net>
+2011-04-06 22:57:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
- * ext/alsa/Makefile.am:
- * ext/cdparanoia/Makefile.am:
- * ext/gio/Makefile.am:
- * ext/gnomevfs/Makefile.am:
- * ext/libvisual/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsptranport: ensure valid int result when parsing ranges
+ Specifically, make sure that the return value of strtol is falling in
+ between the range of G_MININT and G_MAXINT.
+ Fixes #646952.
+
+2011-04-06 16:27:54 +0100 Bastien Nocera <hadess@hadess.net>
+
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-profile: fix unused-but-set-variable warnings with gcc 4.6
+ Top-level profiles don't have restrictions, only stream profiles,
+ so no need to serialise that here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=646925
+
+2011-04-11 14:29:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ tests: dist all struct_*.h files for libs ABI test
+ Should fix distcheck on x86_64.
+
+2011-04-11 15:02:38 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: empty caps have no structure to pick
+
+2011-04-11 10:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstid3tag.c:
+ tag: fix typo in ID3 genres
+ psychadelic -> psychedelic. Spotted by Sébastien Wilmet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=647399
+
+2011-01-27 17:28:51 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * Android.mk:
+ * android/alsa.mk:
+ * android/app.mk:
+ * android/app_plugin.mk:
+ * android/audio.mk:
+ * android/audioconvert.mk:
+ * android/audioresample.mk:
+ * android/audiotestsrc.mk:
+ * android/decodebin.mk:
+ * android/decodebin2.mk:
+ * android/ffmpegcolorspace.mk:
+ * android/gdp.mk:
+ * android/gst-libs/gst/app/gstapp-marshal.c:
+ * android/gst-libs/gst/app/gstapp-marshal.h:
+ * android/gst-libs/gst/audio/audio-enumtypes.c:
+ * android/gst-libs/gst/audio/audio-enumtypes.h:
+ * android/gst-libs/gst/interfaces/interfaces-enumtypes.c:
+ * android/gst-libs/gst/interfaces/interfaces-enumtypes.h:
+ * android/gst-libs/gst/interfaces/interfaces-marshal.c:
+ * android/gst-libs/gst/interfaces/interfaces-marshal.h:
+ * android/gst-libs/gst/pbutils/pbutils-enumtypes.c:
+ * android/gst-libs/gst/pbutils/pbutils-enumtypes.h:
+ * android/gst-libs/gst/rtsp/gstrtsp-enumtypes.c:
+ * android/gst-libs/gst/rtsp/gstrtsp-enumtypes.h:
+ * android/gst-libs/gst/rtsp/gstrtsp-marshal.c:
+ * android/gst-libs/gst/rtsp/gstrtsp-marshal.h:
+ * android/gst-libs/gst/video/video-enumtypes.c:
+ * android/gst-libs/gst/video/video-enumtypes.h:
+ * android/gst/playback/gstplay-marshal.c:
+ * android/gst/playback/gstplay-marshal.h:
+ * android/gst/tcp/gsttcp-enumtypes.c:
+ * android/gst/tcp/gsttcp-enumtypes.h:
+ * android/gst/tcp/gsttcp-marshal.c:
+ * android/gst/tcp/gsttcp-marshal.h:
+ * android/interfaces.mk:
+ * android/netbuffer.mk:
+ * android/pbutils.mk:
+ * android/playbin.mk:
+ * android/queue2.mk:
+ * android/riff.mk:
+ * android/rtp.mk:
+ * android/rtsp.mk:
+ * android/sdp.mk:
+ * android/tag.mk:
+ * android/tcp.mk:
+ * android/typefindfunctions.mk:
+ * android/video.mk:
+ * android/videoscale.mk:
+ * android/videotestsrc.mk:
* ext/ogg/Makefile.am:
- * ext/pango/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
* gst/adder/Makefile.am:
+ * gst/app/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
+ * gst/encoding/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
- * gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
- * sys/v4l/Makefile.am:
- * sys/ximage/Makefile.am:
- * sys/xvimage/Makefile.am:
- Don't install static libs for plugins. Fixes #550851 for base.
-
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tools/Makefile.am:
+ android: make it ready for androgenizer
+ Remove the android/ top dir
+ Fixe the Makefile.am to be androgenized
+ To build gstreamer for android we are now using androgenizer which generates the
+ needed Android.mk files.
+ Androgenizer can be found here:
+ http://git.collabora.co.uk/?p=user/derek/androgenizer.git
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
- Set the default blocksize to -1 because we will then use the configured
- samplesperbuffer to create our output buffer.
+2011-04-09 02:01:08 +0100 Christian Fredrik Kalager Schaller <christian.schaller@collabora.co.uk>
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-plugins-base.spec.in:
+ Add new header file to spec file
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add mappping for the KMVC (Karl Morton's Video) Codec.
+2011-04-08 15:10:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtp: Unref events if the parent element disappeared or has no event handler implemented
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- Don't forget to advance the offset of what we're matching against, else
- we end up in a forever loop.
+2011-01-06 18:20:58 +0100 Ole André Vadla Ravnås <oravnas@cisco.com>
-2008-10-15 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtp: fix pad callbacks so they handle when parent goes away
+ 1) We need to lock and get a strong ref to the parent, if still there.
+ 2) If it has gone away, we need to handle that gracefully.
+ This is necessary in order to safely modify a running pipeline. Has been
+ observed when a streaming thread is doing a buffer_alloc() while an
+ application thread sends an event on a pad further downstream, and from
+ within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
+ while the streaming thread has its buffer_alloc() in progress.
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
- Improve typefinding a bit. If we don't have a Unicode charset
- try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
+2011-03-20 08:59:33 +0100 Havard Graff <havard.graff@tandberg.com>
-2008-10-14 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Make src query MT-safe
+ It is possible that the element might be going down while the event arrives
- * ext/theora/theoradec.c: (theora_dec_decode_buffer):
- Fix build on macosx.
+2011-04-08 15:00:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: Unref events if the parent element disappeared
- Based on patch by: Robin Stocker <robin at nibor dot org>
+2011-03-21 16:03:16 +0100 Havard Graff <havard.graff@tandberg.com>
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (theora_dec_setcaps), (theora_handle_type_packet),
- (theora_dec_decode_buffer), (theora_dec_change_state):
- Parse input caps and make the PAR override the encoded PAR when
- specified by a container. Fixes #555699.
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: make upstream queries and events MT-safe
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-06 16:25:37 +0100 Bastien Nocera <hadess@hadess.net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_change_state):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Add some more G_LIKELY
- Fail when the setcaps function was not called.
-
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps):
- Propagate return value of setcaps.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: Remove unused variables
+ https://bugzilla.gnome.org/show_bug.cgi?id=646924
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-04-07 10:06:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_class_init), (gst_sub_parse_init),
- (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
- (get_next_line), (gst_sub_parse_data_format_autodetect),
- (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
- (gst_subparse_type_find):
- * gst/subparse/gstsubparse.h:
- Add support for UTF16/UTF32 subtitles as long as the first bytes of
- the first buffer contain the BOM. This also adds support for other
- encodings that allow NUL bytes via the encoding property.
- Fixes bugs #552237 and #456788.
+ * gst-libs/gst/video/video.c:
+ video: Fix creation of grayscale caps
+ The endianness was not set correctly before.
+ Fixes bug #646923.
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-04-06 16:11:02 +0200 Robert Swain <robert.swain@collabora.co.uk>
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Don't drop the last byte of image tags if they're not an URI list.
- Fixes bug #556066.
+ * docs/design/part-interlaced-video.txt:
+ docs: Update interlaced video design document
+ The RFF flag is to be reused for buffers in the telecine state to
+ indicate that the buffer contains only unneeded repeated fields that are
+ present in other buffers and as such this buffer can be dropped.
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-25 16:59:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- For looking at the 4th byte we have to get 4 bytes of course
- and not 3.
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: refactor multipass file writing
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-08 14:02:20 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- Improve FLAC-without-headers typefinding by looking at most of the
- frame header and checking if invalid values are used. Should prevent
- quite some false positives compared to the old version which only
- check if the first 14 bits are set.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: minor simplification
+ ... which avoids crashing in the off-chance that structure == NULL.
-2008-10-11 Stefan Kost <ensonic@users.sf.net>
+2011-04-05 18:14:49 +0300 Stefan Kost <ensonic@users.sf.net>
- * sys/xvimage/xvimagesink.c:
- Don't assert on caps==NULL.
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/discoverer.c:
+ tests: add basic unit tests for discoverer
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-08-24 13:14:33 +0200 Pascal Buhler <pascal.buhler@tandberg.com>
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect), (handle_buffer),
- (gst_sub_parse_change_state):
- * gst/subparse/gstsubparse.h:
- * tests/check/elements/subparse.c: (GST_START_TEST):
- Add support for subtitle files with UTF-8 BOM at the beginning
- by simple stripping it from the first line before passing it
- to any parsing code. Fixes bug #555257 and playback of files
- created by Gnome Subtitles.
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk
-2008-10-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-04-05 11:32:52 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_class_init), (gst_audio_test_src_init),
- (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
- (gst_audio_test_src_start), (gst_audio_test_src_stop),
- (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
- (gst_audio_test_src_create):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Define the default property values in the usual place.
- Implement start/stop to reset values correctly.
- Calculate the sample size only once when we negotiate.
- Rename some values to make more sense.
- Keep track of our byte range.
- Add support for pull based scheduling. Disabled for now until we have
- the whole stack working.
- Set the BUFFER_OFFSET correctly.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: don't paint the window black when going to NULL
+ Leave dealing with the appearance of the window when we are not playing to the
+ applications. We anyway want to go to NULL as quickly as possible.
+ Fixes #635800
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-04-04 16:00:30 -0700 David Schleef <ds@schleef.org>
- Based on a patch by: xavierb at gmail dot com
+ * gst-libs/gst/video/video.c:
+ * tests/check/libs/video.c:
+ video: Fix YUV9 and YVU9 again
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- * tests/check/elements/subparse.c: (GST_START_TEST):
- Make the detection of the used subtitle a bit less strict
- for srt subtitles. Fixes bug #555607.
+2011-04-04 23:41:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ tag: fix compiler warning on OSX
+ gstvorbistag.c: In function 'gst_tag_list_from_vorbiscomment_buffer':
+ gstvorbistag.c:371: warning: 'data' may be used uninitialized in this function
- * ext/vorbis/vorbisenc.c:
- (gst_vorbis_enc_buffer_check_discontinuous):
- Fix discontinuity detection which was broken by last commit.
+2011-04-04 23:23:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-09 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/libs/.gitignore:
+ tests: ignore xmpwriter unit test binary
- * configure.ac::
- Require core CVS for ghostpad API additions used by decodebin2.
+2011-04-04 17:21:45 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
-2008-10-08 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: use gst/math-compat.h header.
+ https://bugzilla.gnome.org/show_bug.cgi?id=646744
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Fix debug statements (space between '%' and actual format).
-
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
- Remove bogus assert, the decodepad could have been created inside an
- already existing group.
-
-2008-10-08 Andy Wingo <wingo@pobox.com>
-
- * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
- target instead of setting it.
- (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
- API for a decode pad. The bugfix is that we set the group in
- activate(), not when the pad was created because it might be NULL
- then.
- (gst_decode_group_control_source_pad, gst_decode_group_expose):
- Update to use the API.
-
-2008-10-08 Andy Wingo <wingo@pobox.com>
-
- * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
- be a subclass of GstGhostPad.
- (analyze_new_pad): So, when emitting the signals that determine
- how we do autoplugging, already create the ghost pad and use it as
- the pad in the signal arguments. This allows applications to make
- a connection between the pad passed in e.g. autoplug-continue, and
- the pad passed in new-decoded-pad.
- (connect_pad, expose_pad): Update to receive the ghosted decode
- pad in the args, retargetting it as necessary if we have to plug
- the target pad through a multiqueue.
- (gst_decode_group_control_source_pad): Adapt to receive an
- already-ghosted pad that just needs activation, blocking, and
- drain notification.
- (sort_end_pads): Adapt for decode pads actually being pads.
- (gst_decode_group_expose): Adapt for decode pads actually being
- pads. Rewrite the decode pad names so they appear in order. Adds a
- new error case if we couldn't set the name.
- (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
- logic.
- (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
- New API for the decode pad, needed because we shouldn't do these
- things inside gst_decode_pad_new(), but after.
- (gst_decode_pad_new): Change to actually make the real pad, and
- delay the blocking/drainage bits.
-
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- Patch by: Daniel Drake <dsd at laptop dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
- Unref all buffers when clearing collectpads. Fixes bug #546955.
-
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- Based on a patch by: Klaas <klaas at rivercrew dot net>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
- (gst_vorbis_enc_buffer_check_discontinuous),
- (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
- * ext/vorbis/vorbisenc.h:
- Keep track of the upstream segments and use the running time on that
- segment instead of the buffer timestamp everywhere. Fixes bug #525807.
+2011-04-04 17:23:53 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/xmpwriter.c:
+ tag: Remove constness to silence MS compiler.
+ https://bugzilla.gnome.org/show_bug.cgi?id=646744
- * gst/audioconvert/audioconvert.c: (audio_convert_convert):
- Prevent overflows with big buffer when calculating the size of
- the intermediate buffer by using gst_util_uint64_scale() instead of
- plain arithmetics. Fixes bug #552801.
+2011-04-04 17:23:13 +0200 Haakon Sporsheim <haakon.sporsheim@gmail.com>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: Explicit cast to GThreadFunc to silence MS compiler.
+ https://bugzilla.gnome.org/show_bug.cgi?id=646744
- Patch by: Pavel Zeldin <pzeldin at gmail dot com>
+2011-04-04 15:56:50 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
- (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
- (gst_clock_overlay_init), (gst_clock_overlay_set_property),
- (gst_clock_overlay_get_property):
- * ext/pango/gstclockoverlay.h:
- API: Add ability to specify format for date/time display by
- adding a "time-format" property.
- Fixes bug #554879.
+ * common:
+ Automatic update of common submodule
+ From 1ccbe09 to c3cafe1
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-11 10:41:11 +0100 Trond Andersen <trondand@cisco.com>
- Patch by: Jan Gerber <j at oil21 dot org>
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: fix invalid read in validation of padding in rtcp packet
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add FFV1 fourcc to support playback of FFMPEG lossless video
- in AVI. Fixes bug #555319.
+2011-02-23 10:55:12 +0100 Stian Johansen <stian.johansen@tandberg.com>
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: Add src object lock around call to ringbuffer parse caps.
+ A race was observed between query() and setcaps() where the latter would
+ change the ringbuffer spec while the former was performing operations
+ based this data.
- Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+2011-01-22 23:09:32 +0100 Havard Graff <havard.graff@tandberg.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Implement skew clock slaving. Fixes #552559.
+ baseaudiosrc: protect against ringbuffer disappearing while in a query
+ Observed a case where the src went to null-state during the query,
+ hence the spec pointer was no longer valid, and
+ gst_util_unit64_scale_int crashed (assertion `denom > 0´failed)
+ Add locking to make sure the ringbuffer can't disappear.
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-02-08 18:27:43 +0100 Havard Graff <havard.graff@tandberg.com>
- * gst-libs/gst/audio/multichannel.c:
- * gst-libs/gst/audio/testchannels.c:
- Fix include of config.h
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: don't allow aligning behind the read-segment
+ Given a large enough drift-tolerance, one could end up in a situation
+ where one would keep aligning the written buffers behind the current
+ read-segment position. The result for the reader would be complete
+ silence, possible preceded by very choppy audio.
+ By checking the available headroom, one can determine if there is
+ room to do alignment, or if one should resort to a resync instead to get
+ the pointers back on track.
+ Also refactor the alignment-logic out of the render function for cleaner
+ code.
+
+2011-04-01 13:55:26 -0700 David Schleef <ds@schleef.org>
+
+ * gst/encoding/Makefile.am:
+ * gst/playback/Makefile.am:
+ Remove setting of plugindir from Makefiles
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-23 23:10:51 -0700 David Schleef <ds@schleef.org>
- Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
+ * gst-libs/gst/video/video.c:
+ * tests/check/libs/video.c:
+ video: Fix height calculation for YUV9/YVU9
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
- (print_media), (gst_sdp_message_dump):
- Fix parsing of the c= field containing multicast addresses.
- Fixes #552199.
- Add the connection info to the session or streams.
- Fix parsing of the bandwidth.
- Add debugging for the connections and bandwidths for a media.
- Add debugging for the bandwidth of the session.
+2011-04-01 15:34:30 +0200 Josep Torra <n770galaxy@gmail.com>
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix warning building in mac os x
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_change_state):
- Configure the next seqnum and timestamp in the state change so that they
- can be queried soon after.
+2011-04-01 15:33:42 +0200 Josep Torra <n770galaxy@gmail.com>
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: fix comparison is always false due to limited range of data type
+ Perform calculation in a temp var with enough room as there's guarantee that
+ ret will be able to hold the result for example in _blit_AYUV.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain):
- Improve debugging of the rtptime.
+2011-04-01 12:52:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-05 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Write GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE as METADATA_BLOCK_PICTURE
+ This is the official, standardized way of embedding images into
+ vorbiscomments now.
- * configure.ac:
- Back to development -> 0.10.21.1
+2011-04-01 12:28:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Add support for METADATA_BLOCK_PICTURE tags
+ This is the official, standardized way of embedding pictures
+ inside vorbiscomments now. Parsing code taken from flacparse
+ and slightly changed.
+ Fixes bug #635669.
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
- (plugin_init):
- Add typefinding for FLAC without headers in the beginning.
- This is already handled by flacdec and friends.
+2011-04-01 12:09:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Use g_base64_decode_inplace()
+ Instead of using the GLib base64 decoding functions manually to
+ do inplace base64 decoding. This makes the code easier to understand.
- * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
- (plugin_init):
- Add typefinder for MXF.
+2011-04-01 11:00:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ oggmux: Store the segment directly inside the pad
+ Also initialize it always in TIME format. We require TIME segments
+ in oggmux anyway and drop newsegment events in other formats and
+ assume an open-ended segment starting at 0.
- * tests/icles/Makefile.am:
- Only build test-colorkey if GTK+ is available.
+2011-04-01 10:57:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.21 ===
+ * ext/ogg/gstoggmux.c:
+ oggmux: Reset the segment on flush-stop events and when going back to READY
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+2011-03-03 08:45:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * configure.ac:
- releasing 0.10.21, "We Could Be Wrong"
+ * ext/ogg/gstoggmux.c:
+ oggmux: Use running time instead of timestamps
+ Theora and vorbis use running time (which is correct) for calculating
+ the granulepos for their ogg packets. Oggmux, however, used
+ timestamps to order the received buffers.
+ This patch makes it use the running time to compare buffer times
+ and also to timestamp pushed buffers.
+ Some bits of the code still use timestamps, but they are only
+ used to calculate durations, so it should be fine.
+ https://bugzilla.gnome.org/show_bug.cgi?id=643775
+
+2011-02-16 16:07:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-28 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ oggmux: Keep track of pad's segments
+ https://bugzilla.gnome.org/show_bug.cgi?id=643775
- * configure.ac:
- 0.10.20.4 pre-release
+2011-04-01 10:39:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Add support for xBGR and RGBx
+ Now all RGB variants are supported.
- Patch by: Vincent Penquerc'h <ogg dot k dot ogg dot k at googlemail dot com>
+2011-01-17 21:12:18 -0700 Lane Brooks <dirjud@gmail.com>
- * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
- Set the BOS flag on the BOS packet. Fixes #553244.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Added support for ARGB and other RGB alpha variants
-2008-09-23 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-11 10:34:33 -0700 Lane Brooks <dirjud@gmail.com>
- * gst-libs/gst/rtsp/gstrtspmessage.c:
- (gst_rtsp_message_parse_request),
- (gst_rtsp_message_parse_response):
- Fix the g_return_val_if_fail() statements.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: converted AYUV to use 'A OVER B' alpha compositing
+ 'A OVER B' compositing is explained at
+ http://en.wikipedia.org/wiki/Alpha_compositing.
+ Previously, overlaying text on a transparent background image left the
+ text overlay also transparent. This pipeline shows such an example:
+ gst-launch videotestsrc pattern=white ! video/x-raw-yuv,format=\(fourcc\)AYUV ! alpha alpha=0.0 ! textoverlay text=Testing auto-resize=False font-desc=60px ! videomixer ! ffmpegcolorspace ! autovideosink
+ With this patch, text is composited "OVER" the background image and
+ thus is visible regardless of the alpha of the background image. The
+ overlay in the above pipeline works after applying this patch.
+
+2011-03-28 22:00:25 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-09-22 Michael Smith <msmith@songbirdnest.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ baseaudiosink: arrange for running clock when rendering eos
+ Commit ba2e500bd992d8ad7db0da923801964964835967 ensured to provide
+ a running clock when EOS had finished rendering. However,
+ other measures are needed (and were in place before) to ensure a
+ running clock when EOS still needs rendering (i.e. waiting).
+ So, specifically, re-introduce eos_rendering removed in aforementioned commit,
+ this time as a public variable so subclasses can be aware of the situation.
+ Fixes (part of) #645961.
+ API: GstBaseAudioSink:eos_rendering
- * gst-libs/gst/tag/gsttagdemux.c:
- Fail to activate if there's insufficient data in the file to be usable,
- preventing an assertion fail later. Fixes #552960
+2011-03-31 12:37:32 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-09-15 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/libs/libsabi.c:
+ * tests/check/libs/struct_i386_osx.h:
+ tests: Fixes libsabi for MacOSX/32bit.
+ GStaticRecMutex is 60bytes on macosx/32bit (As opposed to 40).
+ Fixes #644996
- * gst-libs/gst/pbutils/descriptions.c:
- * gst/typefind/gsttypefindfunctions.c:
- Recognise Kate subtitle streams (#550582).
+2011-03-31 10:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-13 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/libs/libsabi.c:
+ * tests/check/libs/struct_x86_64.h:
+ libsabi: Add structure sizes for x86-64
- * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
- Remove trailing comma from enum list, which causes problems
- with -pendantic (#550729).
+2011-03-09 11:51:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ libs: make sure gobject-introspection scanner calls gst_init()
+ Cherry-picked from 0.11, since it's the right thing to do (we
+ now silently rely on various _get_type() working without
+ gst_init() having been called).
- * configure.ac:
- 0.10.20.2 pre-release
+2011-03-30 20:57:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * po/LINGUAS:
- * po/id.po:
- * po/pt_BR.po:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ libs: replace 0.10 with @GST_MAJORMINOR@ in Makefile.am
+ For easier cherry-picking/merging later.
- New translations.
+2011-03-30 15:47:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/gst/typefindfunctions.c:
+ * tests/files/Makefile.am:
+ * tests/files/hls.m3u8:
+ tests: add typefind test for application/x-hls
+ To make sure we don't break detection when we add typefinding
+ for normal m3u8 playlists.
- * gst-libs/gst/interfaces/propertyprobe.c:
- (gst_property_probe_get_properties),
- (gst_property_probe_get_property),
- (gst_property_probe_probe_property),
- (gst_property_probe_probe_property_name),
- (gst_property_probe_needs_probe),
- (gst_property_probe_needs_probe_name),
- (gst_property_probe_get_values),
- (gst_property_probe_get_values_name),
- (gst_property_probe_probe_and_get_values),
- (gst_property_probe_probe_and_get_values_name):
- More sanity checks for our second-favourite interface.
+2011-03-30 15:44:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-05 Stefan Kost <ensonic@users.sf.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: rename type playlist/m3u8 to application/x-hls
+ We should keep playlist/m3u8 available for normal m3u8 playlists,
+ which we we'll likely support some day. Also, we probably don't
+ want this handled like other playlists, so application/* seems
+ more appropriate in this case, even if it's really just a playlist.
- * gst-libs/gst/interfaces/propertyprobe.c:
- Check for NULL pointer, in the hope that this fixes #532864.
+2011-03-30 09:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Fix comment typo and add a link the the HTTP live streaming spec
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- No really, the next release is 0.10.21 (fix Since: tags in docs).
+2011-03-30 09:12:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-04 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Use the DataScanCtx for the m3u8 typefinder
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
- Disable a code path that is now called but causes a deadlock for some
- reason and is unneeded.
+2011-02-14 19:05:09 +0100 Andoni Morales Alastruey <amorales@flumotion.com>
-2008-09-04 Stefan Kost <ensonic@users.sf.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add m3u8 playlists
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add a "draw-border" property that can be set to false to disable
- drawing borders.
-
- * tests/icles/test-colorkey.c:
- * tests/icles/Makefile.am:
- Add new test application for the colorkey handling.
+2011-03-21 15:34:09 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * tests/check/Makefile.am:
+ * tests/check/libs/xmpwriter.c:
+ tagxmpwriter: Add check tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=645167
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
- This will also be fixed for upcoming gst-ffmpeg release so that once
- this release of -base is out, it will work with the latest gst-ffmpeg
- release.
+2011-03-17 15:42:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstxmptag.c:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/xmpwriter.c:
+ * gst-libs/gst/tag/xmpwriter.h:
+ * win32/common/libgsttag.def:
+ tagxmpwriter: Adds a new GstTagXmpWriter interface
+ The GstTagXmpWriter interface is to be implemented on elements that
+ provide xmp serialization. It allows users to select which
+ xmp schemas should be used on serialization.
+ API: GstTagXmpWriter
+ https://bugzilla.gnome.org/show_bug.cgi?id=645167
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Add Truespeech mapping for RIFF formats (AVI/WAV).
- Fixes #550656
+2011-03-18 09:28:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-03 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * gst-libs/gst/tag/tag.h:
+ * win32/common/libgsttag.def:
+ tag: xmp: Add function to list the available schemas
+ Adds a function to list the available schemas in our xmp lib
+ https://bugzilla.gnome.org/show_bug.cgi?id=645167
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
- Fixes #550638.
+2011-03-29 15:41:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/encodebin.c:
+ encodebin: Requesting a pad again now gives a g_return_val_if_fail()
+ Before the behaviour was undefined and implemented differently by elements,
+ now core checks for this (and other problems) and returns NULL and an assertion.
- * configure.ac:
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c:
- * gst/subparse/samiparse.c:
- * tests/check/elements/subparse.c:
- Rework last change, so that we build subparse, but just disable the
- sami parse functionality, if we're configured to not use xml. In the
- tests only the sami test is disabled now.
+2011-03-26 19:36:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggparse.c:
+ oggparse: fix list iteration code
+ Not that it really matters, but let's fix it before someone
+ notices and makes fun of us.
- * configure.ac:
- Disable subparse when xml is disabled. It woundn't work anyway. Fixes
- test runs.
+2011-03-26 12:01:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/libs/.gitignore:
+ tests: ignore new libsabi test binary
- * po/POTFILES.in:
- Add some more files with strings for translation.
+2011-03-26 11:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggparse.c:
+ oggparse: make sure buffer metadata is writable before setting caps on buffers
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c:
- Use new geo location tags from core. Fixes #481169
+2011-03-25 22:14:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-01 Edward Hervey <edward.hervey@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From 193b717 to 1ccbe09
- * tests/check/elements/audioresample.c: (setup_audioresample),
- (fail_unless_perfect_stream), (test_perfect_stream_instance),
- (test_discont_stream_instance):
- Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
- Add debugging for coherence.
+2011-03-25 14:55:52 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-08-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * common:
+ Automatic update of common submodule
+ From b77e2bf to 193b717
- Patch by: Jonathan Matthew <notverysmart gmail com>
+2011-03-25 11:06:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for PDF documents (which is nice to have, since it's a
- common format, but also helps prevent false positives). Fixes #549814.
+ * docs/plugins/Makefile.am:
+ docs: do xrefs for non installed books too
+ Get the xrefs from the builddir for the books in the same package. This fixes
+ the cross references if one does not have the docs already installed.
-2008-08-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-02-25 16:46:29 +0100 Robert Swain <robert.swain@collabora.co.uk>
- * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
- (no_more_pads_cb):
- Fix nasty race where multiple decodebins could start pushing data before
- we manage to configure the sinks, resulting in not-linked errors in
- typical RTSP streaming cases.
+ * docs/design/part-interlaced-video.txt:
+ docs: Add an interlaced video design document
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-25 09:29:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
- Since we now call stop, we trigger this code path that causes a deadlock
- is apparently not needed.
+ * common:
+ Automatic update of common submodule
+ From d8814b6 to b77e2bf
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-25 09:03:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
- (gst_ring_buffer_stop):
- Also allow the case where the ringbuffer was paused when we try to stop
- it so that the basesrc stop function is still called.
+ * common:
+ Automatic update of common submodule
+ From 6aaa286 to d8814b6
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-24 18:48:59 +0200 Stefan Kost <ensonic@users.sf.net>
- Patch by: Mike Ruprecht <cmaiku at gmail dot com>
+ * common:
+ Automatic update of common submodule
+ From 6aec6b9 to 6aaa286
- * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
- Reprobe devices again instead of taking a cached list as new
- devices could've been plugged in. Fixes bug #549062.
+2011-03-24 14:22:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaysink.c:
+ playsink: Update comment about why an audio queue is needed
- Patch by: Alessandro Dessina <alessandro nnva org>
+2011-03-24 14:21:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain):
- Don't add pads and activate them for skeleton streams. These are already
- handled inside oggdemux. Fixes bug #537599.
+ * gst/playback/gstplaysink.c:
+ Revert "playsink: Only add a queue before the audio sink if visualizations are enabled"
+ This reverts commit df886c0622257bb8635e5bd0fc7fc3da20bfc3be.
-2008-08-22 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-24 14:03:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
- Reset variable so that query and convert fail after going back to
- READY. Fixes #548898.
+ * gst/playback/gstplaysink.c:
+ playsink: Only add a queue before the audio sink if visualizations are enabled
+ The queue is not needed otherwise and will add some delay to track
+ switches.
-2008-08-22 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-23 12:42:04 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- If a buffer arrives with a timestamp before the timestamp+duration
- of the previous buffer clip it instead of dropping it completely.
- Slight improvement for the unfixable bug #548913.
+ * tests/check/libs/video.c:
+ tests: video: Uncommenting test
+ Pushed a commented test by accident, uncommenting it.
-2008-08-21 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-23 12:02:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Take the current timestamp instead of timestamp+duration for the offset.
- This offset will later be used for calculating the timestamp and
- otherwise vorbisdec will interpolate timestamps wrong if upstream
- only sends timestamps and no granulepos.
+ * win32/common/libgstvideo.def:
+ video: adds missing function to win32 def
-2008-08-21 Stefan Kost <ensonic@users.sf.net>
+2011-03-23 12:02:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/examples/seek/seek.c:
- Don't crash when having no visualisations.
-
-2008-08-16 David Schleef <ds@schleef.org>
-
- * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
- check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
- Fixes #548065.
+ * gst-libs/gst/video/video.c:
+ video: Getting component offsets without dimensions is fine if it is not YUV
+ This fixes a regression that an assertion would happen if
+ gst_video_get_component_offset would be called with width or
+ height as 0.
+ Calling it with 0 is fine if the format isn't yuv and this
+ was already being used in some other places of video.c
-2008-08-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-03-23 11:13:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- When cleaning up the caps fields also remove "depth" for the same
- reason we remove "width".
+ * tests/check/libs/video.c:
+ tests: video: Add a test for checking rgb caps creation
+ This new test for checking rgb caps creation exposes a regression
-2008-08-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-03-15 14:45:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
- Add Lead H.264 here as well.
+ * gst/playback/gstplaysink.c:
+ playsink: Remember automatically created sinks for future reconfigures
+ Also allow reuse of sink elements in error cases.
-2008-08-14 Julien Moutte <julien@fluendo.com>
+2011-03-16 15:27:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps): Add Lead H.264 variant.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Check if an already existing sink supports the non-raw format too
+ Before we were assuming that a sink will always support all non-raw formats
+ in a single stream.
-2008-08-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-10 19:04:51 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- When not slaved to another clock also subtract the base_time from our
- internal clock time to get the running time.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Check if an element accepts requisite caps before selecting
+ In addition to ensuring that an element we want to select in
+ autoplug-select can enter the READY state, we also now check if it can
+ accept the caps we wish to plug it for. This is handy for sinks that
+ need to perform a probe to figure out whether they can actually handle a
+ given format.
-2008-08-12 David Schleef <ds@schleef.org>
+2011-03-16 15:56:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
- since it has no basis in libtheora.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Set sinks to READY before checking if it accept caps
+ Fixes bug #642732.
-2008-08-12 Stefan Kost <ensonic@users.sf.net>
+2011-03-16 15:56:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/interfaces/propertyprobe.h:
- Remove double "interface" from doc-string.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Always prefer the custom set sink and also set it back to NULL in all cases.
- * gst-libs/gst/interfaces/xoverlay.h:
- Document interface.
+2011-03-17 13:47:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff.c:
- Add basic doc blobs.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Only consider the audio/video sinks in autoplug_continue for the normal uridecodebin
+ Considering them for the subtitle uridecodebin will add audio/video
+ streams that might be in a file used as subtitle file.
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2011-03-22 11:59:40 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/Makefile.am:
- Don't try to build that example anymore.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add gst_video_format_new_template_caps()
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2011-02-24 08:42:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/.cvsignore:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- * gst-libs/gst/audio/make_filter:
- Move audiofiltertemplate to gst-template.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix assertion on caps fixation
+ When fixating caps, from_par should always be initialized
+ with a fixed value.
+ In case the fixation is from src to sink pad it was setting
+ the from par (srcpad par) to a fraction range, this patch initializes
+ it to 1/1, based on the assumption that missing PAR is 1/1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=641952
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2011-03-22 12:44:49 +0100 Luis de Bethencourt <luis@debethencourt.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudiosrc.h:
- More docs and shuffling. What can we do with the hundreds of #defines.
+ * configure.ac:
+ configure.ac: redundant use of AC_MSG_RESULT()
+ cleaned the redundant use of AC_MSG_RESULT() in configure.ac
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
- * gst-libs/gst/audio/audio.h:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/interfaces/propertyprobe.h:
- * gst-libs/gst/tag/gsttagdemux.h:
- Reducing number of dundocumented symbols.
+ * autogen.sh:
+ autogen: wingo signed comment
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2011-03-21 19:22:30 +0100 Fraxinas <andreas.frisch@multimedia-labs.de>
- * gst-libs/gst/audio/audio.c:
- Fix doc comment syntax.
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Fix syntax in Example: Creating a profile
+ https://bugzilla.gnome.org/show_bug.cgi?id=645437
- * gst-libs/gst/interfaces/propertyprobe.c:
- Add more doc-comments and a FIXME: for the signal.
+2011-03-21 18:33:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-08-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Add missing schema creation
+ tiff schema entries were being added to the previous
+ schema (xap) because a new one wasn't being created
+ for it.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
- (gst_ogg_mux_request_new_pad):
- * ext/ogg/gstoggmux.h:
- Don't pretend to support NEWSEGMENT events, instead override the
- GstCollectPads event function to return FALSE on NEWSEGMENT events
- and do the normal work for other events.
-
- This prevents elements like flacenc to seek to the start and rewrite
- some data which then results in a broken Ogg packet.
-
-2008-08-07 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- Patch by: Frederic Crozat <fcrozat@mandriva.org>
-
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
- * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
- * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
- * gst/playback/gstdecodebin.c: (plugin_init):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
- * gst/playback/gstplayback.c: (plugin_init):
- * gst/playback/gstqueue2.c: (plugin_init):
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
- * sys/v4l/gstv4l.c: (plugin_init):
- Make sure gettext returns translations in UTF-8 encoding rather
- than in the current locale encoding (#546822).
-
-2008-08-06 Stefan Kost <ensonic@users.sf.net>
+2011-03-17 21:50:15 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c:
- Add audio/x-qdm for qtdemux.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: Off-by-one error when creating RTP header extensions with a two-byte header
-2008-08-05 Stefan Kost <ensonic@users.sf.net>
+2011-03-16 15:38:31 +0200 Mart Raudsepp <mart.raudsepp@collabora.co.uk>
- * ext/vorbis/vorbisdec.c:
- Do not leak old taglist.
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Clean up alignment docs a bit and remove horiz top alignment enum
-2008-08-04 Stefan Kost <ensonic@users.sf.net>
+2011-02-07 09:13:39 +0200 Mart Raudsepp <leio@gentoo.org>
- * tests/icles/test-scale.c:
- Include <stdlib.h> for atoi().
+ * tests/check/Makefile.am:
+ check: Really fix the linking order of libs/tag
+ Follow-up to commit 5f5c52c, which only fixed the CFLAGS order.
+ Fix the linker order as well.
-2008-08-04 Andy Wingo <wingo@pobox.com>
+2011-03-16 10:19:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
- documentation fix.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: post proper error message if decodebin2/typefind elements are missing
+ Post better error messages in case typefind/decodebin2 are missing or
+ could not be loaded for some reason (e.g. because they inadvertently
+ got blacklisted).
+ https://bugzilla.gnome.org/show_bug.cgi?id=644892
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+2011-03-15 19:47:11 +0100 Blaise Gassend <blaise@suitabletech.com>
- * gst/adder/gstadder.c:
- Cleanup lots of empty lines that came from gst-indent going havoc
- before I added the INDENT_ON/OFF marker some time agao.
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: Store return values of poll functions in a signed integer
+ Negative return values are used for errors and storing
+ them in an unsigned integer will make it impossible to
+ detect the errors.
+ Fixes bug #644845.
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+2011-03-14 19:42:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * gst-libs/gst/riff/riff-read.c:
- Bump requirement to latest core and use new tag for riff formats.
- Needed for #520694.
+ * ext/ogg/gstoggmux.c:
+ oggmux: Increase the seen header packets count when seeing a header packet
+ This fixes muxing of Speex content and possibly other formats where the
+ header detection works by counting the packets.
+ Fixes bug #644745.
-2008-08-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-14 18:35:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/dynamic/Makefile.am:
- * tests/examples/dynamic/codec-select.c: (make_encoder),
- (make_pipeline), (do_switch), (my_bus_callback), (main):
- Add example app that dynamically switches between 3 'encoders'.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add depth and endianness to DTS caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=644208
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-14 11:14:04 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
- Add some more comments.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: use a class wide mutex to work around pango reentrance issues
+ Pango is not reentrant. Use a class wide mutex to protect pange use in
+ gst_text_overlay_render_pangocairo(). This works reliable in contrast to the
+ hack in my previous commit.
+ Fixes Bug #412678
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-14 11:12:53 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
- (gst_video_test_src_create):
- Discard buffers of the wrong size after renegotiation, this is perfectly
- possible with things like capsfilter that could suggest caps changes
- upstream without knowing the size of the buffer.
+ * ext/pango/gsttextoverlay.c:
+ Revert "textoverlay: add a hack to init the pango engine"
+ This reverts commit fee3266056b522cdd34e606b5682553d35eec5a1.
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-03-14 10:09:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-scale.c: (make_pipeline), (main):
- Add dynamic rescaling tests for the new basetransform.
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin2.c:
+ plaback: trim trailing whitespace
-2008-07-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-03-14 10:05:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/audioconvert/Makefile.am:
- Dist recently-added gstfastrandom.h.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: reflow configuring new multiqueue instance
+ Use a single g_object_set to configure the new multiqueue instance. Also don't
+ needlessly set "use-buffering" if it is the default.
-2008-07-30 Edward Hervey <edward.hervey@collabora.co.uk>
+2011-03-04 14:52:01 +0200 Stefan Kost <ensonic@users.sf.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Fix a "may be used uninitialized in this function" which weirdly only
- appears on macosx (?).
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: drop trailing whitespaces
-2008-07-30 Stefan Kost <ensonic@users.sf.net>
+2011-03-04 14:52:28 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/riff/riff-ids.h:
- Adding acid chunk for tempo and loop information.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: add a hack to init the pango engine
+ Layout a single char to pre-create all resources.
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2011-03-12 17:51:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * sys/xvimage/Makefile.am:
- floor() needs linking to $(LIBM).
+ * configure.ac:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/gstlibscpp.cc:
+ tests: add libscpp unit test to make sure g++ likes our library headers
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2011-03-10 14:22:38 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c:
- Aggregate short reads and add some comments and debug logging.
- Fixes #537380
+ * tests/check/elements/encodebin.c:
+ tests: encodebin: Add reuse test case
+ Adds a test case to check if encodebin can be reused
+ https://bugzilla.gnome.org/show_bug.cgi?id=644416
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2011-03-10 14:38:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstplaybasebin.c:
- Fix property doc markup (its not a signal).
+ * gst/encoding/gstencodebin.c:
+ encodebin: Tear down old profiles when setting new ones
+ In NULL/READY, we should be able to switch profiles on encodebin,
+ this patch makes it tear down old profiles when new ones are set
+ if in NULL/READY states
+ https://bugzilla.gnome.org/show_bug.cgi?id=644416
- * sys/xvimage/xvimagesink.c:
- Add since tag for new proeprties (also add sice tags fro the last two
- other additions).
+2010-10-22 14:01:26 +0200 Andoni Morales Alastruey <amorales@flumotion.com>
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: disconnect inactive clients in the select loop too
+ Clients are usually disconnected in the streaming thread if their inactivity
+ is bigger than the timeout. If no new buffers are to be rendered in the sink,
+ these clients will never be disconnected and for that reason it should be
+ handled in the select() loop too.
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add autofill/colorkey properties. Fixes #538656.
+2010-11-03 14:37:07 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2008-07-28 David Schleef <ds@schleef.org>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Don't wait for subtitle streams to preroll
+ Subtitle streams being parse can cause the pipeline to wait indefinitely
+ to PREROLL. This makes subtitle streams got to PAUSED even if no data is
+ available. This should not be a cause for concern as we don't expect to
+ get much data for subtitle streams other than language tags from the
+ container.
+ https://bugzilla.gnome.org/show_bug.cgi?id=632291
- * sys/xvimage/xvimagesink.c:
- Fix rounding errors when converting colorbalance values
- between hardware and object property ranges. Partial
- fix for #537889, however, there still seems to be a small
- drift problem that could be totem's fault.
+2011-03-03 19:14:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: start ringbuffer upon going to PLAYING and already EOS
+ ... otherwise we may end up without running clock in PLAYING.
+ Fixes #636886.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
- Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
- This fixes a critical warning.
+2011-03-04 14:39:45 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: set several properties in one go
+ g_object_set is a varargs function. Save 7 g_obvject_calls (and the overhead of
+ them) by using it accordingly.
- * ext/ogg/gstoggmux.c:
- Allow muxing of CELT into Ogg streams.
+2011-03-02 15:38:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: fix compiler warning on 32-bit systems
+ Mark 64-bit interger constant as such to avoid warnings such as:
+ gsttypefindfunctions.c:2152: error: integer constant is too large for ‘long’ type
- * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
- (plugin_init):
- Add simple typefinder for the CELT codec (www.celt-codec.org).
+2011-02-28 18:52:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-07-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * configure.ac:
+ configure.ac: export plugin description more platform independent
+ Fixes #642504.
- Patch by: Jan Gerber <j at oil21 dot org>
+2011-02-28 18:32:33 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
- Fix calculation of the start time from skeleton streams.
- Fixes bug #530068.
+ * common:
+ Automatic update of common submodule
+ From 1de7f6a to 6aec6b9
-2008-07-24 Stefan Kost <ensonic@users.sf.net>
+2011-02-28 10:10:22 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c:
- Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
+ * tests/check/Makefile.am:
+ * tests/check/libs/libsabi.c:
+ * tests/check/libs/struct_i386.h:
+ tests: add ABI test suite for libs
-2008-07-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-27 09:32:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_setup_dither),
- (gst_audio_quantize_free_dither):
- * gst/audioconvert/gstfastrandom.h:
- Implement a linear congruential generator as pseudo random number
- generator for the dither noise. This is about 2 times faster than
- using GLib's mersenne twister. Also this uses only integer math for
- generating integers while GLib internally uses floating point math.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Only prevent to autoplug the same parser multiple times for the same chain
+ Parsers are the only element class that are not changing the data and
+ could lead to an infinite loop. Other element classes like demuxers,
+ e.g. id3demux, can be used multiple times in a row and sometimes are.
-2008-07-23 Michael Smith <msmith@songbirdnest.com>
+2011-02-26 23:43:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Break the double-factory checking loop immediately if the factory was used already
-2008-07-23 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-02-26 23:39:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't use the same element multiple times in the same chain
+ This is going to lead to an infinite loop of this element and can easily
+ happen with parsers that accept their own src caps on the sinkpad.
- * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
- Use GST_STR_NULL to avoid crashes with libcs that don't
- like NULL strings in printf args (such as the win32 one).
- Fixes #544306.
+2011-02-26 23:20:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Improve detection of raw caps in expose-all-streams=false mode
+ Previously we only checked against the raw caps but we should also
+ check against the return value of autoplug-continue. Additionally fix
+ a thread-safety issue with accessing the raw caps.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
- Oops - set the size of the image used for probing back to 1x1, for
- consistency with ximagesink
+2011-02-25 19:37:07 -0800 David Schleef <ds@schleef.org>
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for r210
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new):
+2011-01-03 11:41:56 +0100 Robert Swain <robert.swain@collabora.co.uk>
- Apparently on Solaris and OS/X (at least), it's not legal to ask the
- X server to attach to a shared memory segment after we've deleted it,
- with the result that MIT-SHM is disabled. Instead, remove it only after
- X succeeds in attaching too.
+ * gst-libs/gst/video/video.h:
+ gstvideo: Add GST_VIDEO_BUFFER_PROGRESSIVE flag
+ Maps to GST_BUFFER_FLAG_MEDIA4. The purpose is to explicitly indicate
+ whether a telecined buffer is progressive or not without having to make
+ assumptions based on previous buffers.
-2008-07-16 David Schleef <ds@schleef.org>
+2011-02-24 20:59:48 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Add 'ticks', a 1/30 second sine wave pulse every second.
+ * tests/check/elements/encodebin.c:
+ encodebin: Fix double unref in unit test
-2008-07-15 David Schleef <ds@schleef.org>
+2011-02-22 14:54:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/video/video.c: Revert ABI change.
+ * tests/check/elements/playbin2.c:
+ checks: add a simple unit test for the source-setup signal
-2008-07-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-22 12:56:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Make it impossible to have NULL caps at the point where we set
- framerate and other things. Also don't return immediately for "3ivd"
- video and let framerate, etc be set. Might fix bug #542508.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2, uridecodebin: add "source-setup" signal
+ Add "source-setup" signal for convenience and discoverability. No need
+ to figure out "notify::source", look up the notify callback signature,
+ then do an g_object_get() to get the source element..
+ https://bugzilla.gnome.org/show_bug.cgi?id=626152
-2008-07-14 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+2011-02-24 16:22:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
- Video format can also be conveniently determined from (many)
- non-fixed caps.
+ * ext/ogg/gstoggmux.c:
+ oggmux: Don't handle GstCollectData as GstObject, use the pad instead
-2008-07-14 Jan Schmidt <thaytan@noraisin.net>
+2011-02-24 16:02:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c:
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreamselector.c:
- First stab at integrating DVD subpicture overlay into
- playbin. Successfully plugs and plays, but the queues need
- shrinking - 3 seconds of video is too much buffering.
+ * tests/check/elements/encodebin.c:
+ encodebin: Fix memory leaks related to request pads
+ Request pads have to be released by the caller and must be
+ unreffed after releasing them.
-2008-07-11 Stefan Kost <ensonic@users.sf.net>
+2011-02-24 15:55:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Remove now obsolete note in the docs.
+ * gst/encoding/gstencodebin.c:
+ encodebin: Return a new reference of the pad for the "request-pad" signal
+ The GObject signal code assumes that the signal handlers return a
+ new reference or copy. Fixes bug #641927.
-2008-07-11 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 20:34:41 -0800 Leo Singer <leo.singer@ligo.org>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-overrides.txt:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- * docs/plugins/gst-plugins-base-plugins.signals:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-cdparanoia.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gdp.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playback.xml:
- * docs/plugins/inspect/plugin-queue2.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-uridecodebin.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml:
- * ext/alsa/gstalsamixer.c:
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasrc.c:
- * ext/gio/gstgiosink.c:
- * ext/gio/gstgiosrc.c:
- * ext/gio/gstgiostreamsink.c:
- * ext/gio/gstgiostreamsrc.c:
- * ext/gnomevfs/gstgnomevfssink.c:
- * ext/gnomevfs/gstgnomevfssrc.c:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/pango/gsttimeoverlay.c:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/theora/theoraparse.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * ext/vorbis/vorbisparse.c:
- * ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/gdp/gstgdpdepay.c:
- * gst/gdp/gstgdppay.c:
- * gst/playback/gstdecodebin2.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gstplaybin2.c:
- * gst/playback/gstqueue2.c:
- * gst/playback/gsturidecodebin.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/videorate/gstvideorate.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- Cleanup Plugin docs. Link to signals and properties. Fix sub-section
- titles. Drop mentining that all our example pipelines are "simple"
- pipelines.
+ adder: Fill in offset_end field of outgoing buffers
+ ... rather than leave it as GST_BUFFER_OFFSET_NONE
+ Fix bug #642942.
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * tests/examples/seek/Makefile.am:
- Fix out of tree build by adding all required CFLAGS.
+2011-02-23 14:31:13 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaysink.c:
+ playsink: release all chains when going to NULL
+ Also fixes #642466.
- * gst/playback/gstdecodebin.c: (add_raw_queue):
- And ref the pad before returning it again when linking to the queue
- failed. Otherwise we will unref the pad twice later and things break.
+2011-02-23 14:29:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaysink.c:
+ playsink: undo state change side effect on error way out
+ ... to avoid subsequent cleanup disposing an element not in NULL state.
- * gst/playback/gstdecodebin.c: (add_raw_queue):
- If linking the raw pad with a queue fails, try it without a queue
- instead of failing completely. This should never happen.
+2011-02-23 10:32:08 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaysink.c:
+ playsink: avoid crashing on the way out when needed chain missing
- Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
+2011-02-22 15:26:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
- Add a queue after a demuxer if the demuxer outputs raw data. This was
- done before only for non-raw data but is required in this case too.
- Fixes bug #540215.
+ * win32/common/libgstvideo.def:
+ win32: update .def file for new libgstvideo API
- decodebin2 doesn't have this issue because all streams of a group
- go through multiqueue.
+2011-02-22 16:41:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-07-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tools/gst-discoverer.c:
+ discoverer: handle desc==NULL
+ It would otherwise be printed as (null) and mess up indentation (no \n).
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+2011-02-08 12:42:32 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Makes libgstsdp compile with mingw32 by defining the right WINVER so
- that getaddrinfo() can be used. Fixes #541358.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Chain dispose() up to parent class
-2008-07-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-02-07 13:04:55 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_class_init), (gst_video_test_src_init),
- (gst_video_test_src_set_property),
- (gst_video_test_src_get_property), (gst_video_test_src_create):
- * gst/videotestsrc/gstvideotestsrc.h:
- Cleanups, use default property values as defines.
- Add property to enable/disable peer buffer allocation.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Keep a ref for the async timeout callback
+ This makes sure we maintain a ref on the discoverer object while the
+ async timeout callback is alive to prevent a potential crash if the
+ object is freed while the callback is pending.
+ https://bugzilla.gnome.org/show_bug.cgi?id=641706
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-07 13:57:39 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
- * tests/check/pipelines/streamheader.c: (streamheader_suite):
- Enable unit tests on PPC again as the bugs are now fixed.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Use g_signal_connect_object instead of g_signal_connect
+ We want to make sure the discoverer object passed to the various
+ callbacks doesn't become invalid if a callback is pending and the object
+ is free'd in the mean time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=641706
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-10 03:22:42 +1100 Parthasarathi Susarla <partha.susarla@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
- Fixes bug #540351.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: detect raw h.263
+ https://bugzilla.gnome.org/show_bug.cgi?id=623846
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-21 15:58:16 +0200 Teemu Katajisto <teemu.katajisto@digia.com>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size):
- Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
- it on other formats. Also adjust the unit size only for that format
- to not include the palette. Fixes bug #540497.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ pbutils: encoding-target: fix error checking in target file loading
+ https://bugzilla.gnome.org/show_bug.cgi?id=642949
-2008-06-29 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 17:55:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c:
- Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
+ * tests/check/elements/videoscale.c:
+ tests: fix videoscale test by ignoring newly-added 64-bit formats
+ They probably fail because ffmpegcolorspace can't handle those formats.
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 18:01:04 +0100 Benjamin Otte <otte@redhat.com>
- * ChangeLog:
- ChangeLog surgery.
-
- * tests/examples/seek/seek.c:
- Move variable into ifdef too.
-
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: Fix copy/paste error in inrospection part of Makefile
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 18:00:36 +0100 Benjamin Otte <otte@redhat.com>
- * tests/examples/seek/seek.c:
- Include config.h and check if we have X. Remove XInitThread(), don't
- think its needed. Fixes: #540334.
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Fix copy/paste error in inrospection part of Makefile
-2008-06-26 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2011-02-21 18:00:02 +0100 Benjamin Otte <otte@redhat.com>
- Patch by: Sam Morris <sam at robots dot org to uk>
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Fix copy/paste error in inrospection part of Makefile
- * gst-libs/gst/interfaces/mixertrack.c:
- (gst_mixer_track_class_init), (gst_mixer_track_get_property),
- (gst_mixer_track_set_property):
- API: Add "index" property to GstMixerTrack to differantiate between
- multiple mixer tracks with the same label.
+2011-02-21 12:40:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Set the "index" property of GstMixerTrack to the index given by ALSA.
- Fixes bug #528299.
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ audiorate: add skip-to-first property
+ API: GstAudioRate::skip-to-first
-2008-06-25 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 12:27:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
- Remove libgstvideo usage. Use gtk_get_option_group instead of
- gtk_init().
+ * gst/videorate/gstvideorate.c:
+ videorate: fix skip-to-first ts setup
+ ... such as avoiding arithmetic mixing counts and ts, although latter
+ would typically be 0 so far.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2011-02-21 12:04:09 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * tests/check/Makefile.am:
- Name the test registry format neutral.
+ * ext/ogg/gstoggmux.c:
+ * gst/adder/gstadder.c:
+ Revert "oggmux,adder: Check if collectpads has been freed"
+ This reverts commit 6d150873e8b4c23d694b0351570de323b1576d76.
+ Depends on a core commit that was reverted.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2011-02-20 23:49:54 -0800 David Schleef <ds@schleef.org>
- * gst/playback/gstqueue2.c:
- Do not double notify. Remove the unsued return value.
+ * ext/ogg/gstoggmux.c:
+ * gst/adder/gstadder.c:
+ oggmux,adder: Check if collectpads has been freed
+ Core now calls release_pad in finalize, which is usually after
+ the collectpads has been unreffed.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2011-02-19 18:50:37 -0800 David Schleef <ds@schleef.org>
- * ext/alsa/gstalsamixer.c:
- Also consider "speaker" as a name for master volume. If that doesn't
- help look for the first non-mono volume control that also has a
- playback switch.
-
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
-
- * ChangeLog:
- Forgot to save the ChangeLog :/
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_fill_borders.c:
+ * gst/videoscale/vs_fill_borders.h:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ videoscale: Add 16-bit-channel support
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2011-02-19 16:41:43 -0800 David Schleef <ds@schleef.org>
- * tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
- Embedd the xwindow. Allow to play multiple files.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add 16-bit-per-channel formats
-2008-06-24 Jan Schmidt <jan.schmidt@sun.com>
+2011-02-19 12:03:17 -0800 David Schleef <ds@schleef.org>
- * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
- (gst_ximagesink_setcaps):
- * sys/ximage/ximagesink.h:
- When the caps change, make sure to re-draw borders in
- force-aspect-ratio=true mode.
- * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
- Don't clear the border_draw flag until we actually draw the border.
- * tests/check/Makefile.am:
- Ignore alsasink/src during the states test too, so it doesn't fail
- when running without access to the sound device.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add ARGB64 and AYUV64
+ 16-bit per channel formats.
-2008-06-22 Stefan Kost <ensonic@users.sf.net>
+2011-02-18 16:26:59 -0800 David Schleef <ds@schleef.org>
- * tests/examples/seek/seek.c:
- Fix crasher when playing a parse-launch line the 2nd time.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add gst_video_format_get_component_depth()
-2008-06-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2011-02-18 13:27:23 -0800 Leo Singer <leo.singer@ligo.org>
- * tests/check/pipelines/oggmux.c:
- Properly ifdef tests to fix compilation.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: each element gets its own instance of GRand, if needed
+ As a result, pipelines that contain multiple instances of audiotestsrc
+ with the 'wave' property set to 'white-noise', 'pink-noise', or
+ 'gaussian-noise' will run much faster, since they won't be competing
+ for access to the global, lock-protected instance of GRand.
+ Fixes bug #642720.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2011-02-18 17:26:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
- Add get-video-pad, get-audio-pad, get-text-pad action signals to
- playbin2. This allows the user to get to the selector's sinkpads, and
- thus inspect a range of things - caps, tags, etc.
+ playbin2: If a sink claims to support ANY caps assume that it only supports the usual raw formats
+ This should be changed again in 0.11, if a sink really claims to support ANY
+ caps it should support everything or provide correct caps.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2011-02-17 18:11:10 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/gstplaybin2.c:
- Use a different constant for the convert-frame signal id.
- Fixes #537009.
+ * gst/encoding/gstencodebin.c:
+ encodebin: Add a audioconverter after the audio resampler.
+ This allows handling non-native-endianness conversion properly.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2011-02-18 14:04:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
- * gst/playback/gstplaysink.c:
- Fix a whole bunch of typos in comments and log statements.
-
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
-
- * sys/xvimage/xvimagesink.c:
- Don't set colour balance values on the Xv port if the user hasn't
- changed them (via properties or the interface). Avoids accumulating
- rounding errors for the common case.
- Partial fix for bug #537889.
-
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
-
- * gst/playback/gstdecodebin2.c:
- Ensure decodebin2 emits 'drained' signal once, and only once, when all
- pads are drained.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
- (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
- Report the encoder latency. Fixes #538232.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
- (notify_source), (activate_group):
- Implement the source property, emit notify when it changes in the
- underlying uridecodebin.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/examples/seek/seek.c: (stop_cb):
- Free and clear the seek element list so that we don't use invalid
- references when seeking after recreating a gst-launch line.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_render):
- Report latency even if we are not live instead of hiding it.
- Take ts-offset and render-delay of the basesink into account when
- scheduling samples.
- Rework the clipping code so that we can take the various offsets into
- account and still do correct clipping.
-
-2008-06-20 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- Bump verion back to devel -> 0.10.20.1
-
-2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Don't increase the size of non-string image buffers by one as this
- might in theory confuse decoders. Still increase it by one for string
- image buffers to append '\0'.
-
-2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
-
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
- Fix a buffer memleak and remove a confusing and wrong debug output.
- Fixes bug #538663.
-
-=== release 0.10.20 ===
-
-2008-06-18 Jan Schmidt <jan.schmidt@sun.com>
+ playbin2: Use gst_pad_accept_caps() instead of intersecting with the getcaps caps
+ This might be faster and more accurate in some cases to detect if a
+ sink supports a format and autoplugging can be stopped.
- * configure.ac:
- releasing 0.10.20, "Here I Go Again"
-
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- 0.10.19.3 pre-release
-
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
-
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Fix build on win32.
- Patch By: David Schleef <ds@schleef.org>
- Fixes: #536874
-
-2008-06-11 Sebastian Dröge <slomo@circular-chaos.org>
+2011-02-18 12:06:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
- (gst_gio_base_src_create):
- * ext/gio/gstgiobasesrc.h:
- Try to read the requested number of bytes, even if the first
- read returns less than requested, until nothing is read anymore
- or we have the requested amount of bytes. This fixes playback of
- files via Samba as Samba only allows to read 64k at once.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Add default handler for autoplug-select
+ uridecodebin proxies this signal and only the first signal handler
+ will ever be called from decodebin2, which is uridecodebin's proxy
+ signal handler.
- Implement a caching algorithm that makes sure that we read at
- least 4k of data every time. Some elements will try to read a few
- bytes, then seek, read again a few bytes and so on and this is
- painfully slow as every operation has to go over DBus if GVfs is
- used as backend.
+2011-02-18 12:02:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Fixes bug #536849 and #536848.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Return NULL from the default autoplug-sort handler
+ ...instead of copying the array. Returning NULL will result
+ in the original factories array to be used and prevents a useless
+ array copy in most use cases.
- * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
- (gst_gio_src_check_get_range):
- Override check_get_range() to blacklist http/https URIs
- and whitelist file URIs. More to be added on demand.
+2011-02-18 12:01:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-05 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Return NULL from the default autoplug-sort handler
+ ...instead of copying the array. Returning NULL will result
+ in the original factories array to be used and prevents a useless
+ array copy in most use cases.
- * configure.ac:
- 0.10.19.2 pre-release
+2011-02-18 12:00:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Update autoplug-* signal docs from decodebin2
+ uridecodebin proxies these signals.
- * win32/common/libgstrtsp.def:
- * win32/common/libgsttag.def:
- Add new API functions to the dll exports
+2011-02-18 11:58:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Michael Smith <msmith@songbirdnest.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Update documentation of the autoplug-* signals
+ Add notes about the behaviour if multiple signal handlers are connected.
+ For most autoplug-* signals only the first signal handler will ever
+ be invoked.
+ Also add to the autoplug-sort docs that the signal handler can return NULL
+ to specify that the order should change and other handlers get the chance
+ to sort the array.
- * gst/playback/gstplaybasebin.c:
- Disconnect signals from decodebins we created before we remove it
- from playbin, to avoid crashes if the decodebin is eventually
- disposed after the playbin itself (possible if the app takes a
- reference on the decodebin).
- Fixes #536521.
+2011-02-18 11:57:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Keep the original factory list if the sort signal handlers returned NULL
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
- (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
- (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
- (h264_video_type_find), (mpeg_video_stream_type_find),
- (dv_type_find), (mmsh_type_find):
- Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
- copy caps for no good reason (this may be desirable to make it easier
- to detect leaks, but then it should probably be done for all caps
- in the typefinder somewhere).
+2011-02-16 20:14:25 +0900 tskd2@yahoo.co.jp <tskd2@yahoo.co.jp>
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: expose "autoplug-sort" signal
+ It is a proxy of the decodebin2's one, and was missing
+ in the previous code.
+ See bug #642433.
- * tests/check/Makefile.am:
- Do not try to run the check tests for subparse unless it has been
- built.
+2011-02-18 10:57:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Use a recursive mutex for the playbin lock
+ This lock is taken when activating a group, which could result in
+ calling the autoplug-continue callback, which also needs this lock
+ to access the sinks.
+ See bug #642174.
- * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
- (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
- Do not try to run a test which requires vorbisenc unless we have
- actually built it.
+2011-02-18 09:36:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Disconnect signal handlers when removing a failed element
+ This prevents crashes later if one of the signals is emitted after the
+ element was removed from decodebin2 already, which can happen in discoverer.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
- (gst_rtsp_connection_clear_auth_params),
- (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Add a couple of missing argument guards.
- Add a way of setting the DSCP for an RTSP connection.
- Add an accessor method for the ip member of GstRTSPConnection as all
- members are supposed to be private.
+2011-02-15 19:23:48 -0800 David Schleef <ds@schleef.org>
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Fix mpeg TS detection
- * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
- Fixed accidental use of IPv4 options for all IPv6 addresses.
+2011-02-04 17:36:40 -0800 David Schleef <ds@schleef.org>
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: move debug category init earlier
- * gst-libs/gst/interfaces/mixertrack.h:
- Document mixer track flags.
+2011-02-03 22:41:23 -0800 David Schleef <ds@schleef.org>
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstoggstream.h:
+ oggparse: better detection of delta unit flag
+
+2011-01-15 18:21:28 -0800 David Schleef <ds@schleef.org>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: Set speed level while running
+
+2011-01-13 15:12:53 -0800 Ralph Giles <giles@thaumus.net>
+
+ * ext/theora/gsttheoraenc.c:
+ Set the theoraenc speed-level property from libtheora's defaults.
+ The speed-level property, which allows callers to trade of encoding
+ quality for speed in the libtheora api, has a version-dependent
+ maximum and default values. Instead of hardcoding the acceptable
+ range for the theoraenc element's presentation of this setting,
+ we query the library directly at class initialization time and
+ set the maximum and default values from that. If the query fails,
+ we fall back to the previous default setting.
+ To keep the values reported by gst-inspect (which I'm told use
+ the spec values from the class) with those available on an\
+ instantiated element, we remove to setting of enc->speed_level
+ from the initializer and instead pass G_PARAM_CONSTRUCT to
+ the property spec flags, asking g_object to set this property
+ when theoraenc objects are constructed.
+ NB in theory the maximum speed-level could depend on the actual
+ video caps. If later versions of libtheoraenc do this, a second
+ call will need to be made from theora_enc_reset to update the
+ property, since this function is mostly useful for realtime
+ adjustment of performance while the pipeline is running.
+
+2011-02-16 11:57:31 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: don't leak parent tags
+
+2011-02-16 11:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: improve logging (and reindent)
+ Add more logging for the tag merging and use the _OBJECT flavour more.
+
+2011-02-15 17:46:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Optimize autoplug-continue handler a bit
+ Don't build merge the caps of all sinks but check them one-by-one
+ until one supports the caps. Also get reffed caps from the sinkpads
+ instead of a writable copy and add debug output if a sink claims to
+ support ANY caps.
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
- Don't set caps on the buffers that contain a copy of the buffer
- including the caps of them resulting in an always increasing refcount
- of the caps and insanely large caps. Instead include a buffer without
- caps in the new caps. Fixes bug #536475.
+2011-02-15 17:24:28 +0100 Akihiro Tsukada <tskd2@yahoo.co.jp>
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix handling of non-raw custom sinks
+ When autoplugging elements in decodebin2, check if
+ the caps are supported by one of the sink before
+ continuing autoplugging.
+ Fixes bug #642174.
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
- Transform a given PAR to a range on the struct with the generic
- height/width instead of the struct with the possibly restricted
- height/width.
+2011-02-15 17:01:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't leak elements that fail to go to PAUSED after being autoplugged
+ Fixes bug #642381.
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
- Prefer the given format if it contains something stricter than [1,MAX]
- for height or width and only put a structure that requires rescaling
- as second. This makes it possible to use videoscale in pipelines where
- the source can actually produce the wanted height/width but usually
- selects a different one from the requested.
+2011-02-13 14:42:14 +0000 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: Don't reset the video quality setting the bitrate
+ libtheora has two encoding modes, CBR, where it tries to hit a target
+ bitrate and VBR where it tries to achieve a target quality.
+ Internally if the target bitrate is set to anything other then 0 the
+ encoding-mode is CBR.
+ This means that the gstreamer element can leave the video_quality
+ setting alone as long as the user is tweaking the bitrate. Which has the
+ nice side-effect that if the user explicitely sets the bitrate to 0
+ (which is actually the default), the quality value doesn't get reset and
+ one ends up encoding VBR at quality-level 0...
- Based on patch by: John Millikin <jmillikin gmail com>
+2011-02-09 12:45:23 +0100 Andoni Morales Alastruey <amorales@flumotion.com>
- * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
- (gst_vorbis_tag_add_coverart):
- Retrieve COVERART tags from vorbis comments (#512333)
+ * gst/gdp/gstgdppay.c:
+ gdppay: ensure buffer's metadata is writable before setting caps
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-02-14 12:52:59 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
- Don't forget to add new enum value here too (should probably use
- glib-mkenums here...).
+ * common:
+ Automatic update of common submodule
+ From f94d739 to 1de7f6a
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-02-10 23:44:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
- * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
- * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
- (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
- (gst_tag_image_data_to_image_buffer):
- Add two utility functions to avoid code duplication (#512333):
- API: add gst_tag_image_data_to_image_buffer()
- API: add gst_tag_list_add_id3_image()
- API: add GST_TAG_IMAGE_TYPE_NONE enum value
+ * gst-plugins-base.doap:
+ doap: update mailing list location
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+2011-02-08 23:58:56 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add gst_audio_check_channel_positions() to the exported symbols.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Use nominal bitrate if bitrate tag is unavailable
+ If the bitrate tag is unavailable, this falls back to the nominal
+ bitrate tag instead, if that is present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=641860
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+2011-02-08 12:31:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions):
- * gst-libs/gst/audio/multichannel.h:
- API: Make gst_audio_check_channel_positions() public.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: caps can be NULL
+ Don't use and unref NULL caps.
- * tests/check/libs/audio.c: (GST_START_TEST):
- Add some simple checks for gst_audio_check_channel_positions().
+2011-02-02 16:49:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: also push cached events downstream when operating in pull mode
+ Otherwise, having 2 tagdemux in a row followed by an element operating in
+ pull mode will make the second tagdemux implictly eat the first tagdemux'
+ tag event(s).
+ Fixes (part of) #641047.
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- minrange and maxrange are scaled according to the frequency
- multiplier.
+2011-01-21 18:10:29 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggmux.c:
+ oggmux: ensure serialnos are unique
+ We do that by checking a newly generated one is not already used in
+ an existing stream, and doing it again if it is.
+ https://bugzilla.gnome.org/show_bug.cgi?id=640211
- * ext/pango/Makefile.am:
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
- (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
- Use gstvideo functions to calculate strides and plane offsets. Fixes
- rendering issue ('ghost' images of the text on the chroma planes)
- with widths or heights that are not multiples of 8 (#506659 and
- probably also #485729).
+2011-02-02 17:30:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
- (main):
- Test with odd height/width too.
+ * ext/ogg/gstoggmux.c:
+ oggmux: free stream map caps when done
-2008-06-02 Sebastian Dröge <slomo@circular-chaos.org>
+2011-02-02 17:23:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_query_duration),
- (gst_adder_query_latency):
- When using gst_element_iterate_pads() one has to unref every pad
- after usage.
+ * ext/ogg/gstoggmux.c:
+ oggmux: keep IN_CAPS flag check for header buffers as fallback
+ In case the ogg mapper doesn't handle all the accepted input formats
+ (although it really should). Saves us error handling for that case
+ though. Also log caps properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=629196
-2008-05-31 Mark Nauwelaerts <mnauw@users.sf.net>
+2011-01-21 16:05:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init):
- Add a gtk-doc chunk for the new properties to have a Since: indication.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ oggmux: use oggstream for less brittleness in recognizing headers
+ Using the IN_CAPS flag for this is brittle, and will fail if either
+ vorbisparse or vorbistag (which is itself based on vorbisparse) is
+ inserted between oggdemux and oggmux. Possibly other elements too
+ (eg, theoraparse, etc).
+ Using oggstream ensures we Get It Right More Often Than Not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=629196
-2008-05-31 Mark Nauwelaerts <mnauw@users.sf.net>
+2011-02-02 15:33:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
- (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
- (gst_base_audio_src_change_state):
- Provide readable actual-buffer-time and actual-latency-time properties
- that reflect the configured ringbuffer values. Fixes #524724.
- API: GstBaseAudioSrc:actual-buffer-time
- API: GstBaseAudioSrc:actual-latency-time
-
-2008-05-30 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
- (gst_basertppayload_change_state):
- Simply converting the running time into an RTP timestamp by scaling it
- based on the clock-rate is good enough for making an RTP timestamp. This
- has the added benefit that we can later on expose a property with the
- RTP timestamp of running time 0, as is needed for RTSP servers to
- generate the response of the PLAY request.
-
-2008-05-30 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix copy-and-paste typo in property docs
- * gst/audioconvert/gstaudioconvert.c:
- (structure_has_fixed_channel_positions),
- (gst_audio_convert_transform_caps):
- Allow up to 11 positioned channels now that audioconvert can handle
- this but add no default positions for > 8 channels.
-
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Add some unit tests for the above change: Test conversion of
- 11 positioned channels to stereo and the other way around, test
- conversion of 15 unpositioned channels in different ways.
-
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-21 10:56:00 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add gst_audio_clock_reset to the list of exported symbols.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ oggmux: do not skip a pageno at start
+ Discontinuities are automatically signalled by oggdemux at the start
+ of a new stream. When oggmux is yet to output actual data pages,
+ do not signal these discontinuities in the ogg stream.
+ This patch may miss some actual discontinuities at the very start of
+ a stream, but avoids the spurious missing pages when encoding happens
+ normally.
+ A better fix might involve finding a way to distinguish between actual
+ data discontinuities and discontinuities merely marking the start of
+ a new stream.
+ Fixes an issue with ogg page numbering (would skip a number for no
+ reason, which then looks like a packet was lost somewhere) when
+ re-muxing an ogg stream, e.g. when re-tagging in rhythmbox.
+ https://bugzilla.gnome.org/show_bug.cgi?id=629196
+
+2011-02-01 15:57:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: clean up property descriptions
+ Remove "This property requires libtheora version >= 1.1" qualifiers
+ from property descriptions. They aren't needed any longer now that
+ we require libtheora >= 1.1.
+
+2010-08-19 22:31:07 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * configure.ac:
+ * gst-libs/gst/tag/gstid3tag.c:
+ id3tag: map the ID3v2 TENC frame to GST_TAG_ENCODED_BY
+ https://bugzilla.gnome.org/show_bug.cgi?id=627268
- * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
- Remove wrong_channels_identification_header unit test as we now
- support 7 (and more channels).
+2011-01-29 20:43:08 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/tcp/gsttcpserversink.c:
+ tcp: use socklen_t where appropriate rather than specific type
+ In particular, fixes Cygwin build where socklen_t is defined as int
+ in line with native win32 api definition.
- * gst/audioconvert/gstchannelmix.c:
- (gst_channel_mix_fill_one_other):
- If mixing left or right to center (or the other way around) only take
- the complete value if we don't already have the original position in
- the source.
+2011-01-29 19:40:23 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmptag: cast argument to isdigit to int
+ ... as that is the specification and fixes compilation on Cygwin:
+ gstxmptaag.c: In function 'read_one_tag':
+ gstxmptag.c:1015: error: array subscript has type 'char'
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_set_structure_channel_positions_list),
- (gst_audio_fixate_channel_positions):
- Allow rear center together with rear left/right and other previously
- conflicting channel positions. The reason why they weren't allowed
- was the channel mixing implementation in audioconvert.
- Also take this into account when fixing channel layouts.
+2011-01-31 18:06:18 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Allow setting channel positions for 1/2 channels when using
- gst_audio_set_structure_channel_position().
+ * gst-libs/gst/app/gstappsink.c:
+ * tests/check/elements/appsink.c:
+ appsink: add buffer fallback in case the application doesn't handle buffer lists
+ We shouldn't assume the application handles buffer lists, for
+ ease-of-use reasons and for backwards compatibility reasons.
- * gst/audioconvert/gstchannelmix.c:
- (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
- (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
- (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
- Major rewrite of the channel mixing.
+2011-01-26 10:32:32 +0800 Cai Yuanqing <Yuanqing.Cai@tieto.com>
- We now allow previously conflicting channel positions to appear
- together (rear center and rear left/right for example).
- Fixes bug #533817.
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: send new-buffer-list signal
+ Send new-buffer-list signal when emit-signals is TRUE
+ https://bugzilla.gnome.org/show_bug.cgi?id=640607
- Rework the way channels are mixed together to take more possible
- channel positions into account, properly mix from/to side channels
- and don't assume that either center, left&right or nothing of a
- specific position is available anymore.
+2011-01-20 16:25:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Adjust unit tests with non-standard 1/2 channel layouts to the more
- correct new behaviour.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: also add https to buffer protocols
+ HTTPS also needs buffering.
- Add a unit test for 5.1->Stereo downmixing.
+2011-01-30 15:40:53 +0200 Felipe Contreras <felipe.contreras@nokia.com>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: trivial cleanups
+ It seems these stuff was neglected from commmit d8942e2.
+ Signed-off-by: Felipe Contreras <felipe.contreras@nokia.com>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
- Add sane defaults for the 7 and 8 channel layouts as those are
- undefined in the Vorbis spec. Use NONE channel layouts when decoding
- more than 8 channels instead of erroring out. Fixes bug #535356.
+2011-01-27 15:26:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * win32/common/config.h:
+ win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink
+ https://bugzilla.gnome.org/show_bug.cgi?id=640705
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/theora/theoraparse.c:
- Add theoraparse to the docs and fix some docs.
+2011-01-27 12:32:35 +0100 Philippe Normand <pnormand@igalia.com>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: register H264 typefinder with H264 caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=640709
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
- Fix EOS condition and track addition check, the track.end sector is
- included in the track. Fixes #533265.
+2011-01-26 12:16:58 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/encoding/gststreamsplitter.c:
+ streamsplitter: release pending events refs
+ Unref pending events when disposing the streamsplitter.
+ Also refactor a little to replace a for with a g_list_foreach
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+2011-01-26 15:42:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
- (gst_video_rate_flush_prev), (gst_video_rate_event),
- (gst_video_rate_chain):
- * gst/videorate/gstvideorate.h:
- React (more) to NEWSEGMENT
- Small adjustment in timestamp calculation to prevent mismatches
- Fixes #435633.
+ * tests/check/Makefile.am:
+ tests: don't run encodebin test if vorbis or theora plugins aren't available
-2008-05-28 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-26 09:07:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
- Initialise error to NULL as we should.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for degas images
-2008-05-28 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-26 09:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_query_duration),
- (gst_adder_query_latency), (gst_adder_query):
- Implement latency query.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: use image/x-icon as media type for ICON files
+ That's what we've been using so far (e.g. gdkpixbufdec).
-2008-05-27 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-18 10:20:29 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/adder/gstadder.c: (gst_adder_query_duration):
- Correctly resync the iterator if gst_iterator_next() returns
- GST_ITERATOR_RESYNC.
+ * tests/examples/snapshot/snapshot.c:
+ snapshot: use a keyframe seek
+ One would usualy get good quality snapshots quickly. The exact seek position
+ does not really matter.
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-17 23:13:29 +0200 Stefan Kost <ensonic@users.sf.net>
- * win32/vs6/libgstpbutils.dsp:
- Add pbutils-enumtypes.c to sources (#518037).
+ * tests/examples/snapshot/snapshot.c:
+ snapshot: add a newline to the usage and error output
-2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-25 18:03:23 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
- (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
- * gst-libs/gst/audio/gstaudioclock.h:
- Add method to inform the clock that the time starts from 0 again. We use
- this info to calculate a clock offset so that the time we report in
- internal_time is monotonically increasing, as required by the clock base
- class. Fixes #521761.
- API: GstAudioClock::gst_audio_clock_reset()
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add comment and whitespace trimming
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Reset reported time when we (re)create the ringbuffer.
+2011-01-12 14:03:12 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for x-annodex
- * ext/alsa/gstalsamixertrack.c:
- (gst_alsa_mixer_track_update_alsa_capabilities):
- Make sure playback volumes aren't accidentally overwritten by
- capture volumes if an alsa mixer track has both playback and
- capture capabilities: we create two GstMixerTracks in that
- case, so make sure we query only the alsa capabilities that
- refer to the type of GstMixerTrack we created from the dual
- capability alsa element. Should fix issues with Audigy2 sound
- cards (#518082).
+2011-01-25 13:39:25 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add typefinder for DEGAS images
+ This fixes at least one DEGAS image from being misdetected as DTS audio.
+ https://bugzilla.gnome.org/show_bug.cgi?id=625129
- * tests/check/pipelines/oggmux.c: (test_pipeline):
- Don't use deprecated function.
+2011-01-21 14:56:28 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggmux.c:
+ oggmux: cleanup
+ Remove a pointless string concatentation, and fix an off-by-one in
+ packetno in a log.
+ https://bugzilla.gnome.org/show_bug.cgi?id=640189
- * gst/playback/gstdecodebin2.c:
- (gst_decode_group_control_source_pad), (gst_decode_group_expose):
- Check for NULL cases and log them, creating ghostpads can, for example,
- fail when the pad returns wrong caps.
+2011-01-24 11:45:21 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * gst/playback/gstplaybin2.c: (perform_eos):
- When pushing out the EOS event, collect the return value and warn when
- something failed.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add detection for windows icon files to get them out of the way
+ Some of them can otherwise be misdetected for MPEG audio.
+ https://bugzilla.gnome.org/show_bug.cgi?id=620364
-2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-17 15:11:15 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for DVCPRO.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Remove dead code
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-11 15:10:42 +0800 Yang Xichuan <xichuan.yang@tieto.com>
- * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
- Change default scaling method from nearest-neighbour to bilinear.
+ * ext/ogg/gstoggparse.c:
+ oggparse: Make gst_ogg_parse_submit_buffer() safe
+ By not passing zero-sized buffers to ogg_sync_buffer()
+ and checking the return values of libogg functions.
+ Fixes bug #639136.
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-11 18:18:34 +0100 Lane Brooks <dirjud@gmail.com>
- * tests/check/libs/video.c:
- More checks.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Add support for vertical center alignment
+ Fixes bug #639159.
-2008-05-25 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-24 15:21:10 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (parser_state_init),
- (gst_sub_parse_format_autodetect), (handle_buffer):
- * gst/subparse/gstsubparse.h:
- * tests/check/elements/subparse.c: (test_tmplayer_style3b):
- Limit duration to a maximum of five seconds for tmplayer format where
- we can guess the duration only from the timestamp of the next line of
- text. We don't want to show a text for eternities just because nothing
- else is being said for a while.
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Move static variable to local function
+ Variable was being written to and could cause crashes
+ if multiple elements were parsing xmp at the same time.
+ Moving it to local scope solves the problem.
-2008-05-23 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-24 18:27:30 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_handle_sink_event),
- (gst_base_rtp_depayload_push_full),
- (gst_base_rtp_depayload_change_state):
- Check sequence numbers, mark input buffers with a discont flag for the
- subclass when we detected a gap, drop duplicate buffers. We do this
- because one can use the element without a jitterbuffer in front and we
- don't want to feed the subclasses invalid or reordered data.
- Do an error when the subclass did not provide a process function instead
- of crashing.
- Some other small cleanups.
-
-2008-05-22 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
- May just as well use the precalculated uvstride here.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for video/x-camstudio
-2008-05-22 Jan Schmidt <jan.schmidt@sun.com>
+2011-01-24 00:00:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-overrides.txt:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
+ * configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
- * docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggdemux.h:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- * gst/audioconvert/audioconvert.c:
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.h:
- * gst/gdp/gstgdpdepay.h:
- * gst/gdp/gstgdppay.h:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstdecodebin2.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gstplaybin2.c:
- * gst/playback/gsturidecodebin.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.h:
-
- Add some documentation comments, and some new headers to be scanned.
- Rename some internal enum declarations (audioconvert's DitherType and
- NoiseShapingType, GstUnitType from the TCP elements) to match the
- documented GObject type names so that the docs pick them up.
- Name the playbin2 docs markups properly so they get picked up. They'll
- need renaming back when/if playbin2 becomes playbin.
-
- 100% symbol coverage for the plugin docs, booya.
-
-2008-05-22 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
-
- * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
- Fix generation of NV12/NV21 frames. Fixes bug #532454.
-
-2008-05-22 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
-
- * gst/playback/gstdecodebin.c: (remove_fakesink):
- Lock the fakesink before setting the state to NULL and removing it from
- the bin so that a concurrent state change cannot interfere.
- Fixes #534331.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
-
- * docs/Makefile.am:
- Fix installing plugin documentation when gtk-doc is disabled.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
-
- * gst-libs/gst/rtsp/Makefile.am:
- Distribute, don't install md5.h
-
-2008-05-21 Julien Moutte <julien@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
- instead of SOL_IP, works on more platforms.
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
- arguments.
-
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/vorbis/vorbisdec.c:
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
- Some debug and comment fixes.
-
- * tests/examples/dynamic/addstream.c: (main):
- Fix , to ;
-
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- * gst/playback/decodetest.c: (new_decoded_pad_cb):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
- (cleanup_decodebin):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
- (connect_element), (gst_decode_group_control_demuxer_pad):
- * gst/playback/gstplaybasebin.c: (queue_remove_probe),
- (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
- (mute_group_type):
- * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
- (gst_play_bin_set_property), (handoff), (gen_video_element),
- (gen_text_element), (gen_audio_element), (gen_vis_element),
- (remove_sinks), (add_sink), (setup_sinks):
- * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
- * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
- (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
- (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
- (gen_video_chain), (gen_text_chain), (gen_audio_chain),
- (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
- (gst_play_sink_request_pad):
- * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (cb_newpad):
- * gst/playback/test6.c: (new_decoded_pad_cb):
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- * tests/check/elements/audiorate.c: (test_injector_chain),
- (do_perfect_stream_test):
- * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/elements/gnomevfssink.c:
- * tests/check/elements/textoverlay.c:
- (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
- * tests/check/elements/videotestsrc.c: (GST_START_TEST):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- * tests/check/pipelines/oggmux.c: (test_pipeline):
- * tests/check/pipelines/streamheader.c: (GST_START_TEST):
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
- * tests/examples/seek/scrubby.c: (make_wav_pipeline):
- * tests/examples/seek/seek.c: (make_mod_pipeline),
- (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
- (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline),
- (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
- (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
- (update_fill), (msg_buffering):
- Don't use bad gst_element_get_pad().
-
-2008-05-21 Stefan Kost <ensonic@users.sf.net>
-
- * gst-libs/gst/riff/riff-media.c:
- Fix wrong method name in docs. Fix calculation of strf fields for
- broken mulaw/alaw.
-
- * gst-libs/gst/riff/riff-read.c:
- Whitespace fix and removing double ';'.
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.32 ===
- * docs/design/part-playbin2.txt:
- Add some leftover doc.
+2011-01-21 10:50:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.32
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
- Fix copy & paste error in last commit.
+2011-01-18 10:45:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.31.4 pre-releases
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
- Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
- other channel positions when source has SIDE channels and dest doesn't
- or the other way around.
+2011-01-18 10:44:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ docs: update docs
- Patch by: Henrik Eriksson <henriken at axis dot com>
+2011-01-18 10:40:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
- (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
- (gst_multi_fd_sink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Add support for DSCP QOS. Fixes #469933.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * tests/check/libs/profile.c:
+ encoding-target: change keyfile header to 'GStreamer Encoding Target'
+ which is more in line with other files such as .desktop files.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-18 01:06:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Add another test that checks if conversion between standard 1 and 2
- channel layouts with and without positions set is working.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ pbutils: don't assume LC_MESSAGES is always defined, also check for ENABLE_NLS
+ Should fix build with mingw32 build bot again.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-18 00:09:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions):
- Allow non-standard 2 channel layouts.
-
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Add some tests for converting and remapping non-standard 1 and 2
- channel layouts.
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * win32/common/libgstapp.def:
+ app: export gst_app_stream_type_get_type()
+ API: gst_app_stream_type_get_type()
+ API: GST_TYPE_APP_STREAM_TYPE
+ https://bugzilla.gnome.org/show_bug.cgi?id=639747
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-17 23:59:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstchannelmix.c:
- (gst_channel_mix_fill_normalize):
- Prevent division by zero if the channel mix matrix contains only
- zeroes.
+ * gst-libs/gst/app/gstappbuffer.c:
+ app: make GstAppBuffer get_type() function thread-safe
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-18 01:09:53 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Drop new stream tags once preroll is done
+ This makes sure we do not touch the stream taglist once the pipeline has
+ been prerolled. Adding of stream tags happens in the pad event probe
+ which runs in a different thread from discoverer stream processing, so
+ modifying the tag list while discoverer might be processing it can
+ sometimes cause a crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=639778
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
- Close a buffer memory leak. Fixes bug #534071.
+2011-01-17 15:30:08 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Validate timeouts before processing them
+ This avoids a race where the timeout callback is scheduled to run but we
+ get sufficient information to finish discovery before actually getting
+ around to executing the callback. See the documentation of
+ g_source_is_destroyed() for more details.
+ https://bugzilla.gnome.org/show_bug.cgi?id=639730
- * gst-libs/gst/rtsp/gstrtsptransport.h:
- Make the GstRTSPTransport struct members public as there are no
- setters/getters and it's supposed to be changed directly.
- Fixes bug #533087.
+2011-01-18 00:08:32 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Make sure we call _stop() before being freed
+ This ensures that everything is properly cleaned up before the
+ GstDiscoverer object is freed. Specifically, it makes sure that we've
+ removed the async timeout callback before freeing the object to avoid a
+ potential crash later on.
+ https://bugzilla.gnome.org/show_bug.cgi?id=639755
- * gst/adder/gstadder.c:
- Adder also doesn't support audio/x-raw-int with width!=depth so don't
- claim this on the pad template caps.
+2011-01-16 14:55:46 -0800 David Schleef <ds@schleef.org>
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/gdp/gstgdppay.c:
+ gdppay: make newsegment buffer metadata writable
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_sync_latency):
- We can only use our optimal calibration if we prerolled before the
- latency expired.
+2011-01-16 16:46:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-20 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/pbutils/encoding-target.c:
+ pbutils: save localised strings properly when writing encoding targets to a file
+ Use LC_MESSAGES rather than LC_ALL. Save/load description as untranslated string
+ when using an English language locale. Strip locale information to the language,
+ so we don't save keys like description[fr_FR.UTF-8]=...
+ https://bugzilla.gnome.org/show_bug.cgi?id=638860
- * configure.ac:
- Require core CVS for GstBaseSrc buffer caps setting magic.
+2011-01-13 13:59:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: set framed=false on DTS caps
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Fix logic in last commit.
+2011-01-12 17:51:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ docs: add some more Since: markers for new encoding-profile API
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Passthrough the channel positions if the number of output channels is
- the same as the number of input channels, the input had a channel
- layout and downstream requests no special one. We did this already for
- > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
-
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
- (gst_gnome_vfs_src_finalize),
- (gst_gnome_vfs_src_received_headers_callback),
- (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
- * ext/gnomevfs/gstgnomevfssrc.h:
- Set the ICY caps on the srcpad from where they get picked up by the base
- class now and set on the outgoing buffers.
+2011-01-12 15:51:52 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
- BaseSrc now sets the caps on outgoing buffers automatically.
+ * configure.ac:
+ configure: require gobject-introspection >= 0.9.12
+ Earlier versions don't honour the -L/--library-path option,
+ which we need. See commit 4d0ccdad in gobject-introspection git.
+ Should "fix" build on lucid/maverick build bots.
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-11 19:19:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_resample_slaving),
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
- (gst_base_audio_sink_async_play),
- (gst_base_audio_sink_change_state):
- Change the way in which the ringbuffer is started when dealing with a
- slaved clock and latency. We now sync to the clock until we reach
- upstream latency before starting the ringbuffer. This has the effect
- that we can accurately align the master and slave clocks and let the
- rate correction code take care of the initial drift or rounding errors
- instead of leaving them uncorrected with the old approach.
-
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.31.3 pre-release
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Correctly set the default channel positions when converting to 8
- channels.
+2011-01-11 18:59:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * po/da.po:
+ * po/gl.po:
+ * po/pt_BR.po:
+ po: update translations
- * configure.ac:
- Error out if we don't have the required version of core.
+2011-01-11 14:41:53 +0000 Bastien Nocera <hadess@hadess.net>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ examples: allow building with newer GTK+
+ GtkFunction is gone, and there's no update policies for
+ GtkRanges any more (but the default was continuous anyway,
+ so no need to set it to that mode explicitly).
+ https://bugzilla.gnome.org/show_bug.cgi?id=639215
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
- Use data scan helper in aac typefinder and stop scanning
- for headers when we've found a type. Also fix potential invalid
- memory access when calculating the frame length.
+2011-01-11 14:59:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/pbutils/Makefile.am:
+ gobject-introspection: pass --library-path as well to make it find the right libgstreamer
+ Makes things work again properly in uninstalled setups (and
+ presumably in installed setups where GStreamer is installed
+ into a non-standard prefix). Requires fixes from core git.
+ https://bugzilla.gnome.org/show_bug.cgi?id=639039
- * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
- (mpeg_sys_is_valid_pack):
- Don't modify scan context when we return FALSE in ensure_data, so
- it's possible to continue scanning, and we don't end up with a NULL
- data pointer and a positive size, which might bite us the next time
- we're called. Small constification.
+2011-01-11 14:52:51 +0000 Byeong-ryeol Kim <brofkims@gmail.com>
-2008-05-16 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/Makefile.am:
+ gobject-introspection: fix issue when gold linker is used
+ Need to pass libgstreamer-0.10 explicitly to linker, since we're
+ calling gst_init(), which in turn is needed because the encoding
+ target get_type() function calls gst_value_register().
+ https://bugzilla.gnome.org/show_bug.cgi?id=639039
- * gst/adder/gstadder.c:
- Adder doesn't support 24 bit samples so don't claim it supports them
- in the pad template caps.
+2011-01-11 15:49:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From e572c87 to f94d739
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain):
- Validate the RTP packet before further processing it. It's just too
- dangerous to accept random packets and people are not forced to use a
- jitterbuffer or session manager to filter out the bad packets.
+2011-01-10 16:35:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_set_extension_data),
- (gst_rtp_buffer_get_payload_subbuffer):
- Small cleanups.
- When setting extension data in a buffer that is too small, we fail and
- we should not set the extension bit.
- Change GST_WARNINGS into g_warning because they really are
- programming errors.
+ * common:
+ Automatic update of common submodule
+ From ccbaa85 to e572c87
- * tests/check/libs/rtp.c: (GST_START_TEST):
- Catch the g_warnings now in the unit tests and that fact that failing to
- set extension data left the extension bit untouched.
+2011-01-10 14:53:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * common:
+ Automatic update of common submodule
+ From 46445ad to ccbaa85
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Revert previous change which made basetransform handle buffer_alloc
- and which breaks things badly in the non-passthrough case since it
- returned buffers with a different (ie. sometimes smaller) size than
- the size requested.
+2011-01-10 15:55:26 +0800 Yang Xichuan <xichuan.yang@tieto.com>
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: remove outdated comment
+ https://bugzilla.gnome.org/show_bug.cgi?id=639121
- Patch by: Bernard B <b-gnome at largestprime dot net>
+2011-01-08 02:16:19 +0000 Koop Mast <kwm@FreeBSD.org>
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
- Fix seqnum compare function for bordercase values and fix the docs
- again. Fixes #533075.
+ * configure.ac:
+ configure: fix bash-ism
+ https://bugzilla.gnome.org/show_bug.cgi?id=638961
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add a testcase for seqnum compare function.
+2011-01-08 02:10:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-14 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ gobject-introspection: use same PKG_CONFIG_PATH for g-ir-compiler as for g-ir-scanner
+ Make sure to use the PKG_CONFIG_PATH set at configure time instead of
+ just relying on an env-var set one. This makes sure both g-ir-compiler
+ and g-ir-scanner use the same PKG_CONFIG_PATH for determining include
+ paths etc.
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_class_init):
- Correctly declare the supported endianness on the pad templates
- and check for correct endianness in the set caps function. Adder
- only supports native endianness.
- Also use gst_element_class_set_details_simple().
+2011-01-08 01:12:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-14 Stefan Kost <ensonic@users.sf.net>
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-app.pc.in:
+ * pkgconfig/gstreamer-audio-uninstalled.pc.in:
+ * pkgconfig/gstreamer-audio.pc.in:
+ * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
+ * pkgconfig/gstreamer-cdda.pc.in:
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft.pc.in:
+ * pkgconfig/gstreamer-floatcast.pc.in:
+ * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
+ * pkgconfig/gstreamer-interfaces.pc.in:
+ * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
+ * pkgconfig/gstreamer-netbuffer.pc.in:
+ * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
+ * pkgconfig/gstreamer-pbutils.pc.in:
+ * pkgconfig/gstreamer-riff-uninstalled.pc.in:
+ * pkgconfig/gstreamer-riff.pc.in:
+ * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtp.pc.in:
+ * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp.pc.in:
+ * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-sdp.pc.in:
+ * pkgconfig/gstreamer-tag-uninstalled.pc.in:
+ * pkgconfig/gstreamer-tag.pc.in:
+ * pkgconfig/gstreamer-video-uninstalled.pc.in:
+ * pkgconfig/gstreamer-video.pc.in:
+ pkg-config: add girdir and typelibdir variables to .pc files
+ We need them when building gir and typelib files for
+ libraries that depend on these, such as gst-rtsp-server
+ for example, in an uninstalled setup.
- * sys/xvimage/xvimagesink.c:
- Better debug logging in port value handling. Merging separate port
- value loops into one.
+2011-01-07 12:50:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/video-enumtypes.c:
+ 0.10.31.2 pre-release
- Patch by: Hannes Bistry <hannesb at gmx dot de>
+2011-01-07 13:04:11 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
- * gst/tcp/gsttcpserversink.c:
- (gst_tcp_server_sink_handle_server_read),
- (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
- Fix regression in clientsrc because we did not add the fd to the poll
- set anymore. Fixes #532364.
- Do some cleanups here and there.
+ * gst/encoding/gstencodebin.c:
+ * gst/encoding/gstencodebin.h:
+ encodebin: Add missing-plugin support
+ https://bugzilla.gnome.org/show_bug.cgi?id=638903
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-07 12:51:11 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
- Use correct marshallers. GstCaps are a boxed type and no GObject
- subclass.
+ * gst/encoding/gstencodebin.c:
+ encodebin: Extend documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=638901
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-07 00:43:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstrtsp.def:
- Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
- symbols.
+ * tests/check/Makefile.am:
+ tests: never disable g_assert() and cast checks for the unit tests
+ The unit tests are riddled with g_assert() and friends, sometimes
+ containing functional code like set_state() calls in them even
+ (looking at you, pipeline/capsfilter-renegotiation). Make sure we
+ don't disable assert and cast checks for the unit tests even if
+ this has been specified for the rest of the code base, e.g. via
+ --disable-glib-asserts.
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-06 23:17:12 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * win32/common/libgstpbutils.def:
+ win32: udpate pbutils .def file for API change
- * tests/check/elements/audioresample.c:
- (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
- (live_switch_push), (GST_START_TEST):
- Add unit test for the latest basetransform negotiation changes.
- See bug #526768.
+2011-01-06 23:13:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ docs: update docs
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-06 23:13:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
- Fix nv12<->nv21 conversion if stride is larger than width.
+ * po/fi.po:
+ * po/ru.po:
+ po: update translations
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-06 23:08:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Jan Gerber <j at oil21 dot org>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: make text property controllable too
+ Because we can, and because it's the most interesting one
+ to control really, after xpos/ypos.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
- (gst_ogg_pad_parse_skeleton_fisbone):
- * ext/ogg/gstoggdemux.h:
- Parse presentation time from skeleton streams and use it as offset
- for the timestamps. Fixes bug #530068.
+2011-01-06 23:01:20 +0000 Lane Brooks <dirjud@gmail.com>
-2008-05-12 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/pango/Makefile.am:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: make some properties controllable
+ https://bugzilla.gnome.org/show_bug.cgi?id=638859
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
- Revert previous patch that attempted to more accurately calculate the
- initial offset between master and slave clock. The best thing we can do
- in general is take the time of both clocks as the diff since we don't
- know when the actual preroll happened.
+2011-01-06 20:37:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-11 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/libs/.gitignore:
+ tests: ignore new rtsp test binary
- * gst-libs/gst/pbutils/install-plugins.c:
- Fix docs: type and missing word.
+2011-01-05 15:54:15 -0800 David Schleef <ds@schleef.org>
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: ignore header pages when looking for keyframe
+ This was causing keyframe_granule to be set to 0 for all streams
+ when seeking to the beginning of the stream, i.e., at the
+ beginning of playback. Fixes #619778.
- * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
- Don't do lots of 4-byte peeks, but use the 'new' data scan helper
- for this instead; don't check if we've found enough markers after
- each and every step, it's enough to do that only if we've actually
- found a new marker.
- Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
+2010-12-29 15:27:44 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggstream.c:
+ oggstream: when the last keyframe position is not known, do not use -1
+ Instead, use either 0 or 1, depending on bitstream version, which give
+ the correct result for streams which aren't cut off at start.
+ This allows that function to not return negative granpos.
+ https://bugzilla.gnome.org/show_bug.cgi?id=638276
- * gst/typefind/gsttypefindfunctions.c:
- (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
- (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
- (mpeg_video_stream_type_find):
- Move scan helper thingy to the beginning of the file so we can use
- it in other typefind functions. Rename it to something more
- generic. Also improve handling of things towards the end of the
- typefind data: peek as much as we can if we know the size of the
- data, rather than just min_size.
+2011-01-06 17:57:41 +0000 christian schaller <christian.schaller@collabora.co.uk>
+
+ * gst-plugins-base.spec.in:
+ Update spec file with discoverer and encodebinchanges
-2008-05-09 Jan Schmidt <jan.schmidt@sun.com>
+2011-01-05 15:53:09 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/interfaces/colorbalance.c:
- * gst-libs/gst/interfaces/colorbalance.h:
- * gst-libs/gst/interfaces/colorbalancechannel.c:
- * gst-libs/gst/interfaces/colorbalancechannel.h:
- * gst-libs/gst/interfaces/tuner.c:
- * gst-libs/gst/interfaces/tunerchannel.c:
- * gst-libs/gst/interfaces/tunerchannel.h:
- * gst-libs/gst/interfaces/tunernorm.c:
- * gst-libs/gst/interfaces/tunernorm.h:
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Document the GstTuner and GstColorBalance interfaces, and some
- other random API functions that needed it. 70% symbol coverage, woo.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ discoverer: Documentation updates
+ Some cosmetic changes and expands on some bits of the documentation to
+ make it more newbie-friendly.
-2008-05-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2011-01-06 13:08:53 +0100 Robert Swain <robert.swain@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
- Choose to allocate one less segment but require one additional segment
- as latency.
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ videorate: Fix behaviour for frame rate cap changes
+ The outgoing buffer timestamp is calculated by scaling an output buffer
+ count by the src pad frame rate caps. If these caps change, we need to
+ reset the count and work from a new base timestamp. The new output
+ buffer timestamp is then the count scaled by the new caps values added
+ onto the base timestamp.
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
- No need to increment the number of segments in the source.
+2011-01-06 08:47:04 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (clock_convert_external),
- (gst_base_audio_sink_resample_slaving),
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
- (gst_base_audio_sink_async_play):
- Remove adding latency when returning the internal time while subtracting
- it again when we use the value a little later.
- When calculating the end timestamp, we are making a rounding error
- with the current algorithm. Ensure that we don't accumulate these
- rounding errors when aligning samples by not resampling at all if we
- don't need to. Fixes #419351.
- Make the initial calibration of the clock slaving a little more
- predictable and accurate. Also handle the case where we don't do
- clock slaving.
-
-2008-05-09 Sebastian Dröge <slomo@circular-chaos.org>
-
- Based on a patch by:
- Björn Benderius <bjoern dot benderius at axis dot com>
+ * tools/gst-discoverer.c:
+ tools: Improve pretty-printing of tags
+ Avoids escaping strings for nothing and printing out useless buffer contents.
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Add conversions from/to NV12 and NV21 and conversions between those
- two formats. Fixes bug #532166.
+2011-01-06 08:46:42 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-05-08 Edward Hervey <edward.hervey@collabora.co.uk>
+ * tools/gst-discoverer.c:
+ tools: don't leak the GMainLoop
- * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
- Abort the h264 typefinding as soon as _peek() doesn't return anything,
- which happens for example with files smaller than 128kb.
+2011-01-06 00:28:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/pbutils/encoding-target.c:
+ pbutils: config.h include should come before all other includes
- Patch by: Wouter Cloetens <zombie at e2big dot org>
+2011-01-05 22:02:35 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_create), (md5_digest_to_hex_string),
- (auth_digest_compute_hex_urp), (auth_digest_compute_response),
- (add_auth_header), (gst_rtsp_connection_free),
- (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
- (gst_rtsp_connection_set_auth_param),
- (gst_rtsp_connection_clear_auth_params):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Add Digest authorization support for RTSP connections. See #532065.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * gst/encoding/gstencodebin.c:
+ * tests/check/libs/profile.c:
+ * tests/examples/encoding/encoding.c:
+ encoding: encoding_profile_get_output_caps => _get_input_caps
+ Makes more sense name-wise
- * gst-libs/gst/rtsp/md5.c:
- * gst-libs/gst/rtsp/md5.h:
- Yeap, another md5 implementation until we can depend on a glib that has
- support for it.
+2011-01-05 20:40:39 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-05-08 Sebastian Dröge <slomo@circular-chaos.org>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: Add various new symbols
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+2011-01-05 01:50:34 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Let audioresample use the buffer allocation of basetransform instead
- of it's own stuff.
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-profile: Minor documentation updates
- * tests/check/elements/audioresample.c: (alloc_only_48000),
- (GST_START_TEST), (audioresample_suite):
- Add unit test for the recent basetransform bugfix, where upstream
- changes caps to something that can't be passed through anymore.
+2011-01-03 19:07:45 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-05-07 Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Give a better usage example
- * win32/common/config.h.in:
- Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
- use the real thing than having "???" unconditionally.
+2011-01-03 18:52:00 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-05-07 Wim Taymans <wim.taymans@collabora.co.uk>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ * tests/check/libs/profile.c:
+ * win32/common/libgstpbutils.def:
+ encoding-target: Fixup loading/saving methods
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_query):
- Report the latency with the new seglatency parameter.
+2011-01-03 18:51:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
- (gst_ring_buffer_acquire):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add new field to the ringbufferspec to specify the expected latency
- between the underlying device read/write pointer, this is needed
- when writing sinks that sit a little closer to the hardware.
- Add some more docs for other fields.
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ encoding-target: more docs cleanups
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-03 16:07:49 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_transform_ip):
- Return NOT_NEGOTIATED if we didn't set a process function yet for some
- reason instead of crashing later. Might fix bug #509125.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * tests/check/libs/profile.c:
+ encoding-target: Change target suffix to .gep
+ Along with a bunch of other internal cleanups
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-03 13:21:26 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ encoding-target: Add more docs regarding categories
- * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps),
- (structure_has_fixed_channel_positions),
- (gst_audio_convert_transform_caps):
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
- Add support for more than 8 channels and NONE channel layouts. For
- more than 8 channels no channel conversion is supported yet, only
- format conversions are supported. Fixes bug #398033.
+2011-01-03 13:20:19 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST), (audioconvert_suite):
- Add some unit tests by Tim for checking the NONE channel layouts
- and more than 8 channels and add some more unit tests for channel
- conversions.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ * tests/check/libs/profile.c:
+ * win32/common/libgstpbutils.def:
+ encoding-target: Add API for list all categories and targets
+ API: gst_encoding_list_available_categories
+ API: gst_encoding_list_all_targets
-2008-05-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-22 18:18:00 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (connect_pad):
- When autoplugging fails, set the element back to NULL before
- unreffing it.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * tests/check/libs/profile.c:
+ * win32/common/libgstpbutils.def:
+ encoding-profile: Add convenience method to find a profile
+ API: gst_encoding_profile_find
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-22 18:16:33 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add gst_base_audio_src_[sg]et_slave_method() to the exported
- symbols.
+ * configure.ac:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ * tests/check/libs/profile.c:
+ encoding-target: Implement save/load feature
+ Fixes #637735
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-22 11:41:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_sync),
- (end_sami_element), (characters_sami):
- Remove trailing, leading and double whitespaces.
- Correctly timestamp buffers and output the last buffer too.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ * tests/check/libs/profile.c:
+ * win32/common/libgstpbutils.def:
+ encoding-target: Add method to get a profile by name
+ API: gst_encoding_target_get_profile
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add a simple unit test for SAMI parsing.
+2011-01-05 19:30:50 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/encoding/gstencodebin.c:
+ encodebin: Convert to new GstElementClass::request_new_pad_full vmethod
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+2011-01-05 15:31:09 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_sync),
- (start_sami_element), (end_sami_element), (characters_sami),
- (sami_context_reset):
- Only output characters inside the "sync" elements. There could be
- other elements like "style" that have some content but should
- not be printed. Fixes bug #467911.
+ * gst-libs/gst/pbutils/pbutils.h:
+ pbutils: Don't forget to include the encoding headers
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-05 12:02:02 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (set_audio_mute),
- (set_active_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (playbin_set_audio_mute):
- Allow setting -1 as current-audio to mute the current audio stream,
- similar to what is done for subtitles. Fixes bug #342294.
+ * gst-libs/gst/video/video.c:
+ video: Fix uninitialized variables
+ reported by macosx gcc
-2008-05-05 Edward Hervey <edward.hervey at collabora co uk>
+2010-12-07 14:59:46 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- It's SorensOn and not SorensEn.
+ * gst-libs/gst/pbutils/codec-utils.c:
+ codec-utils: Minor documentation changes
-2008-05-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+2011-01-02 15:48:47 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Fix description of video/x-flash-video.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add stream-format to h264 caps
-2008-05-04 Sebastian Dröge <slomo@circular-chaos.org>
+2011-01-02 17:21:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
- * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
- * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
- Remove some unused code.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: default to enable-last-buffer=FALSE for audio sinks
+ There isn't really any good reason to get the last buffer from an
+ audio sink, so don't make the sink keep it around unnecessarily.
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_free_noise_shaping):
- Don't return before freeing the noise shaping history.
+2010-12-31 12:14:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * configure.ac:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstinputselector.h:
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gstplaybin2.c:
+ playbin2: use input-selector from core instead of internal copy
- * tests/check/elements/subparse.c: (do_test),
- (test_tmplayer_style3b), (subparse_suite):
- Add unit test for the tmplayer variant from bug #530962.
+2010-12-31 01:24:50 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ tests: add input-selector-test and output-selector-test
+ Moved from gst-plugins-bad into -base, becasue it uses videotestsrc
+ and other elements from -base, so it can't be in core.
+
+2010-11-24 12:22:01 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/icles/output-selector-test.c:
+ output-selector-test: don't hardcode videosinks and use more colorspace conv.
+ Use autovideosink instead of hardcoded sinks. Use an additional colorspace
+ converter between videotestsrc and timeoverlay.
+
+2009-10-27 11:51:05 -0700 Michael Smith <msmith@songbirdnest.com>
+
+ * tests/icles/output-selector-test.c:
+ tests: Remove executable bits from non-executable files.
+
+2009-02-24 16:33:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/icles/input-selector-test.c:
+ tests: move examples directory to tests/examples as in every other GStreamer module
+
+2008-06-19 13:18:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests: Use BOILERPLATE macro and update output-selector test to the latest api changes.
+ Original commit message from CVS:
+ * gst/selector/gstoutputselector.c:
+ * tests/icles/output-selector-test.c:
+ Use BOILERPLATE macro and update test to the latest api changes.
+
+2008-02-07 13:48:20 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/icles/output-selector-test.c: Add a fixme comment.
+ Original commit message from CVS:
+ * gst/multifile/gstmultifilesink.c:
+ Add a fixme comment.
+ * gst/selector/gstoutputselector.c:
+ Fix same leak as in input-selector.
+ * tests/icles/output-selector-test.c:
+ Improve the test.
+
+2008-01-29 07:38:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Replace the switch plugin with the selector plugin. Add output-selector as the opposite of input-selector (was switc...
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * docs/plugins/gst-plugins-bad-plugins.args:
+ * docs/plugins/gst-plugins-bad-plugins.hierarchy:
+ * docs/plugins/gst-plugins-bad-plugins.interfaces:
+ * docs/plugins/gst-plugins-bad-plugins.signals:
+ * docs/plugins/inspect/plugin-metadata.xml:
+ * docs/plugins/inspect/plugin-selector.xml:
+ * docs/plugins/inspect/plugin-soundtouch.xml:
+ * docs/plugins/inspect/plugin-switch.xml:
+ * gst/selector/.cvsignore:
+ * gst/selector/Makefile.am:
+ * gst/selector/gstinputselector.c:
+ * gst/selector/gstinputselector.h:
+ * gst/selector/gstoutputselector.c:
+ * gst/selector/gstoutputselector.h:
+ * gst/selector/gstselector-marshal.list:
+ * gst/selector/gstselector.c:
+ * gst/selector/selector.vcproj:
+ * gst/switch/.cvsignore:
+ * gst/switch/Makefile.am:
+ * gst/switch/gstswitch-marshal.list:
+ * gst/switch/gstswitch.c:
+ * gst/switch/gstswitch.h:
+ * gst/switch/switch.vcproj:
+ * tests/icles/.cvsignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/output-selector-test.c:
+ Replace the switch plugin with the selector plugin. Add output-
+ selector as the opposite of input-selectoo (was switch). Add a test
+ for output-selector. Add docs for the elements. The vcproj needs
+ update. Fixes #500142.
+
+2010-12-30 18:08:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (handle_buffer),
- (gst_sub_parse_sink_event):
- * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
- (tmplayer_parse_line):
- Fix parsing of tmplayer subtitle variant where every single line contains
- text and there isn't an empty line after each line to determine the
- duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
- making sure that we push out the last line of text without a duration if
- there's still text left in the buffer at the end.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ baseaudiopay: fix timestamps on buffer lists
+ Fix the outgoing timestamps and RTP timestamps on outgoing buffers when using
+ buffer lists.
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-12-29 22:36:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (feed_textbuf):
- Fix detection of discontinuities based on the buffer offset (doesn't work
- so well if no buffer offset is set) and also check for the DISCONT buffer
- flag. This keeps the parser state from being reset after each buffer in
- the unit test.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: assume EBML files without doctype are matroska
+ https://bugzilla.gnome.org/show_bug.cgi?id=638019
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-12-29 12:53:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
- Further fine-tuning: don't absolutely require sequence or GOP headers
- (as introduced in the previous commit), but adjust the typefind
- probabilities returned accordingly if we don't see them. Also make sure
- picture header and first slice are somewhat close to each other (which
- is not perfect but still better than requiring a fixed offset or having
- no limit at all).
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: only keep last valid timestamp
+ Fixes #634397
+
+2010-10-13 17:09:13 +0200 Andoni Morales Alastruey <amorales@flumotion.com>
+
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ multifdsink: add first and last buffer's timestamp to the stats
+
+2010-12-29 11:51:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ ogg: fix typo in comment
+
+2010-12-28 17:39:58 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/ogg/gstoggstream.c:
+ oggstream: fix interpretation of Theora granule position
+ The offset part of the granpos is not a sign of the newer encoding.
+ Use the version number instead.
+ This fixes the criticals thrown by theoraparse, and (at last) the
+ remaining part of #553244.
+
+2010-11-25 17:01:04 +0100 Havard Graff <havard.graff@tandberg.com>
-2008-05-02 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: protect against ringbuffer disappearing while in a query
+ Observed a case where the sink went to null-state during the query,
+ hence the ringbuffer-pointer was NULL, causing a crash.
+ Moving the ringbuffer-check code until after the query, and hold the
+ lock during the check and while using the spec-values. It should not matter
+ to the query wether the ringbuffer is present or not, and it actually
+ gets a time bit more time to get the ringbuffer set up in this case!
+ Fixes #635231
+
+2010-12-28 19:39:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_sink_setcaps),
- (gst_basertppayload_sink_getcaps):
- Rename the setcaps/getcaps function internally to make it clear that
- they are called for the sink pad.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: handle pads that are not added yet
+ Don't try to stream data on pads that are not added yet. This happens while we
+ discover the different streams.
-2008-05-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-28 11:41:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init),
- (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
- (gst_base_rtp_depayload_packet_lost),
- (gst_base_rtp_depayload_set_gst_timestamp):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Catch packet-lost events from the jitterbuffer and convert them into a
- vmethod call (lost-packet) so that depayloaders can do something smart.
- Also add a default packet-lost function that sends out a segment update
- to the decoders.
+ basedepay: fix refcounting issue
+ Make sure that when _make_writable() returns a new buffer, we actually push that
+ one instead of the old one.
-2008-05-02 Stefan Kost <ensonic@users.sf.net>
+2010-12-25 15:22:42 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * gst/playback/test4.c:
- * gst/playback/test5.c:
- * gst/playback/test6.c:
- * gst/playback/test7.c:
- Also include config.h when relying on defines from it. Fixes the
- build. Its been a please to serve :)
+ * ext/ogg/gstoggstream.c:
+ oggstream: implement tag extraction for Kate streams
+ This will mainly allow Totem to know the language of those streams,
+ so the subtitle selection menu gets properly filled out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=638005
-2008-05-02 Thijs Vermeir <thijsvermeir@gmail.com>
+2010-12-26 17:29:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
- (paint_setup_NV21), (paint_hline_NV12_NV21):
- Add support for NV12 and NV21 in videotestsrc
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for DVB subtitle caps
-2008-05-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-23 17:18:17 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c:
- * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
- * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
- (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
- (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
- (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
- (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
- (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
- (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
- (vs_image_scale_linear_RGB555):
- Support 1x1 images as input and output as for example the BBC HQ new
- streams have 1x1 GIFs in the playlists for some reason.
-
-2008-05-01 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
- (try_to_link_1):
- If we can't activate one of the decoders we plugged in (such as,
- say, musepackdec) for some reason (it might not support push mode,
- for example), remove any pad probes that close_pad_link() might
- have set up. This makes sure we later don't try to remove a probe
- for a pad that doesn't exist any longer, and avoids nast warnings
- and probably other things too.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: set headers on caps
+ This will allow switching from one stream to another without having to send
+ the headers for the new stream again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=637927
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-12-22 15:29:56 -0800 David Schleef <ds@schleef.org>
- * gst/typefind/gsttypefindfunctions.c:
- (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
- (plugin_init):
- Rework mpeg video stream typefinding a bit more: make sure sequence,
- GOP, picture and slice headers appear in the order they should and
- that we've in fact at least had one of each; fix picture header
- detection; decouple picture and slice header check - don't assume
- they're at a fixed offset, there may be extra data in between. Also,
- announce varying degrees of probability depending on what we found
- exactly (multiple pictures, at least one picture, just sequence and
- GOP headers). Finally, in _ensure_data(), take into account that we
- might be typefinding smaller amounts of data, such as the first
- buffer of a stream, so fall back to the minimum size needed as long
- as that's available, instead of erroring out if there's less than
- 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
- fuzzed file from #399342 as valid.
+ * ext/ogg/gstoggstream.c:
+ oggstream: Fix parsing of theora size
-2008-04-30 Michael Smith <msmith@songbirdnest.com>
+2010-12-22 19:06:56 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * ext/theora/theoradec.c:
- Cool kids don't divide by zero.
- Treat PAR of x:0 as 1:1.
- Fixes #530719.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't use gst_pad_alloc_buffer()
+ allocate buffers using gst_buffer_new_and_alloc() instead of
+ gst_pad_alloc_buffer_and_set_caps(), as the first one will
+ cause the pad to block, and we don't want that since that will
+ prevent subsequent pads from being fed if a block occurs at
+ start, when all pads must be fed for playback to start.
+ This fixes autoplugging of the tiger element and other things.
+ https://bugzilla.gnome.org/show_bug.cgi?id=637822
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-12-22 18:12:14 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
- (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
- (mpeg_video_stream_type_find):
- Refactor a bit: use context structure to track parsing offset and
- size of available data and make the code a bit clearer. Fixes bad
- memory access in #356937.
+ * gst/encoding/gstencodebin.c:
+ encodebin: Also use "Formatter"s for container formats
-2008-04-28 Michael Smith <msmith@songbirdnest.com>
+2010-12-22 18:19:48 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/test4.c:
- * gst/playback/test5.c:
- * gst/playback/test6.c:
- * gst/tcp/gstmultifdsink.c:
- Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
- is defined.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ encoding-target: Fix typo
-2008-04-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-22 10:32:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Clarify some docs.
-
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
- (gst_base_audio_src_class_init), (gst_base_audio_src_init),
- (gst_base_audio_src_set_slave_method),
- (gst_base_audio_src_get_slave_method),
- (gst_base_audio_src_set_property),
- (gst_base_audio_src_get_property), (gst_base_audio_src_create):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- Add property and methods for selecting the clock slave method in the
- source, like in the sink.
- We only implement "none" and "re-timestamp" for now.
- API: gst_base_audio_src_set_slave_method()
- API: gst_base_audio_src_get_slave_method()
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Fix unitialized data warning
+ Fixes a valgrind warning on jifmux tests on -bad caused by
+ unitialized bytes.
+ Fixes #637758
-2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-22 13:56:12 +0100 Alessandro Decina <alessandro.d@gmail.com>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Fix the docs about the seqnum compare function, it returns a difference.
+ * gst/encoding/gstencodebin.c:
+ encodebin: minor fix in error handling.
+ Don't call gst_bin_remove (bin, <invalid pointer>).
-2008-04-24 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-12-21 18:51:29 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * ext/alsa/gstalsadeviceprobe.c:
- (gst_alsa_get_device_list): Don't return before freeing up
- the allocated structures.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ pbutils: More gtk-doc annotations
-2008-04-24 Stefan Kost <ensonic@users.sf.net>
+2010-12-21 10:26:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Remove obsolete streaminfo code and fix a leak. Fixes #529546
+ * gst/playback/gstplaybin2.c:
+ playbin2: delay stream-changed messages
+ https://bugzilla.gnome.org/show_bug.cgi?id=637586
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-12-21 16:33:50 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * ext/ogg/gstoggdemux.c:
- Revert the event part, that should not go in.
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * tests/check/libs/profile.c:
+ encoding-target: Ensure target names and categories are valid
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-12-21 15:11:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c:
- Don't leak GstPluginFeatures when filtering.
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ depay: update some docs
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-12-21 15:02:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Add some logging for cases when grabbing the xv failed.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ rtpdepayloade: add support for getting events
+ Add support for intercepting sink events in the depayloader by adding a new
+ vmethod.
-2008-04-21 David Schleef <ds@schleef.org>
+2010-12-21 13:37:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c:
- Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
- packet. Should conform to what we currently think is the
- final Ogg/Dirac muxing spec.
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: keep timestamps when no decoded output
+ Keep track of the timestamps even when we didn't generate decodable output.
-2008-04-21 David Schleef <ds@schleef.org>
+2010-12-21 13:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Fix typo that causes the overlay keying color to bright green
- on a 16-bit display. Dark grey good. Bright green bad.
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: avoid using invalid timestamps
-2008-04-21 Stefan Kost <ensonic@users.sf.net>
+2010-12-21 10:41:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfsuri.c:
- Add FIXME comment about using uri-list for source and sink.
+ * tests/examples/seek/seek.c:
+ seek: don't pause for live buffering messages
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-20 18:29:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
- GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
- vaargs functions to gint. Otherwise the fractions will get 0 set
- instead of the correct value on big endian systems. Fixes bug #529018.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: use RTP base time when invalid timestamps
+ When we have an invalid running-time (because we clipped, for example) use the
+ RTP base time for timestamping instead of generating wrong RTP timestamps.
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-20 18:28:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssink.c:
- (gst_gnome_vfs_sink_uri_get_protocols):
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnome_vfs_src_uri_get_protocols):
- * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
- (gst_gnomevfs_get_supported_uris):
- Get the list of supported URI schemes in a threadsafe way and use the
- same list for the source and sink.
-
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/gio/gstgio.c: (_internal_get_supported_protocols),
- (gst_gio_get_supported_protocols):
- Don't generate a new supported protocols list on each call but cache
- it. It's supposed to be static anyway, this way we only leak it once
- per process.
-
- * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
- (gst_gio_sink_class_init), (gst_gio_sink_finalize),
- (gst_gio_sink_set_property), (gst_gio_sink_get_property),
- (gst_gio_sink_start):
- * ext/gio/gstgiosink.h:
- * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
- (gst_gio_src_class_init), (gst_gio_src_finalize),
- (gst_gio_src_set_property), (gst_gio_src_get_property),
- (gst_gio_src_start):
- * ext/gio/gstgiosrc.h:
- API: Add "file" properties where one can set a GFile as
- source/destination.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppayload: copy applied rate to segment
+ Use set_segment_full to copy all segment values to the segment structure.
- Add locking to the properties and use
- gst_element_class_set_details_simple() instead of a static
- GstElementDetails struct.
+2010-12-21 13:09:34 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/check/elements/encodebin.c:
+ * tests/check/libs/profile.c:
+ tests: Update container-less profile checks
- * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
- (plugin_init):
- Add "mpp" and "mp+" as possible extensions for MusePack files.
+2010-12-21 13:08:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- Add typefinding for MusePack StreamVersion 8 files and include the
- stream version in the caps.
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ encoding-profile: Add guard against profiles without format
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-21 13:07:27 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtppayloads.c:
- (gst_rtp_payload_info_for_name):
- Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
+ * gst/encoding/gstencodebin.c:
+ encodebin: Fix usage of non-container profiles
-2008-04-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-17 16:10:53 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * configure.ac:
- Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
- (NB: this only affects compilation of some of the examples).
- Remove some configure.ac cruft that's not needed any longer.
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ docs: Update for videoscale class changes
-2008-04-18 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-12-20 17:46:48 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Don't validate the payload if there isn't any.
- Fixes #525915
+ * common:
+ Automatic update of common submodule
+ From 169462a to 46445ad
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-19 13:41:22 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
- Use g_atomic_int_set() instead of gst_atomic_int_set().
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ gstdiscoverer: Don't leak tags
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-19 13:22:23 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Return NULL instead of a gchar * array with one NULL element if we
- don't get any supported URI schemes from GIO.
+ * tools/gst-discoverer.c:
+ gst-discoverer: show global tags by default
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-12-19 09:53:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Remove cpp style commented old code.
+ * tests/check/libs/rtsp.c:
+ rtsp: Fix memory leaks in the gst_rtsp_url_decode_path_components() unit tests
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-12-18 20:47:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin2.c:
- Fix signal docs.
+ * tests/examples/encoding/Makefile.am:
+ examples: Fix encodebin example CFLAGS and LDFLAGS
+ Previously it would only succeed to link if a new enough
+ libgstpbutils-0.10 was installed in the default library
+ search path.
-2008-04-14 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-17 14:16:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
- (gst_text_overlay_init):
- Fix textoverlay unit test again by making the supposed default
- value for the wait-text property the actual default value.
- Also fix Since: tag for new property.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: implement packet duration query for kate streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=637519
-2008-04-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-17 19:06:27 -0600 Rob Clark <rob@ti.com>
- * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
- (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
- (gst_video_format_get_pixel_stride),
- (gst_video_format_get_component_width),
- (gst_video_format_get_component_height),
- (gst_video_format_get_component_offset), (gst_video_format_get_size),
- (gst_video_format_convert):
- Add guards to these functions to ensure sane input values.
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * gst/encoding/gstencodebin.c:
+ fix compile errors on macosx
+ with i686-apple-darwin10-gcc-4.2.1:
+ encoding-profile.h:134: warning: type qualifiers ignored on function return type
+ encoding-profile.c:240: warning: type qualifiers ignored on function return type
+ gstencodebin.c: In function 'next_unused_stream_profile':
+ gstencodebin.c:454: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
+ gstencodebin.c:464: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
- * tests/check/libs/video.c:
- Fix unit test not to create caps with width=0 and height=0.
+2010-12-17 00:49:26 -0800 Leo Singer <leo.singer@ligo.org>
-2008-04-11 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: corrected buffer duration calculation to account for nonzero initial timestamp
+ Since we calculate timestamps by:
+ timestamp = t0 + (out samples) / (out rate)
+ and durations by:
+ duration = ((out samples) + (processed samples)) / (out rate) - timestamp
+ if t0 is nonzero, this would simplify to
+ duration = t0 + (processed samples) / (out rate).
+ This duration is too large by the amount t0. We should have done:
+ duration = t0 + ((out samples) + (processed samples)) / (out rate) - timestamp
+ so that
+ duration = (processed samples) / (out rate).
+
+2010-12-16 20:40:33 -0800 Leo Singer <leo.singer@ligo.org>
- * docs/design/draft-keyframe-force.txt:
- Fix typo.
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: changed num_gap_samples, num_nongap_samples from guint32 to guint64 so that gaps of greater than or equal to 2^32 samples do not cause integer overflow
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_handle_src_query):
- Set buffering mode in the messages.
- Set buffering percent in the query.
+2010-12-16 20:38:31 -0800 Leo Singer <leo.singer@ligo.org>
- * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
- (do_stream_buffering), (do_download_buffering), (msg_buffering):
- Do some more fancy things based on the buffering method in use.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: push half a history length, instead of a full history length, at end-of-stream so that output segment and input segment have same duration
+
+2010-12-16 20:34:13 -0800 Leo Singer <leo.singer@ligo.org>
+
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: renamed count_gap, count_nongap to more descriptive num_gap_samples, num_nongap_samples
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-16 20:32:07 -0800 Leo Singer <leo.singer@ligo.org>
- * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
- (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
- (msg_buffering), (main):
- Add basic download reports to seek using the new buffering API.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: replaced void* with gpointer
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-16 20:30:24 -0800 Leo Singer <leo.singer@ligo.org>
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
- (gst_queue_src_checkgetrange_function):
- Include extra buffering stats in the buffering message.
- Implement BUFFERING query.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: initial filter transient discarded; unit tests passing
- * gst/playback/gsturidecodebin.c: (do_async_start),
- (do_async_done), (type_found), (setup_streaming), (setup_source),
- (gst_uri_decode_bin_change_state):
- Only add decodebin2 when the type is found in streaming mode.
- Make uridecodebin async to PAUSED even when we don't have decodebin2
- added yet.
+2010-12-16 20:09:58 -0800 Leo Singer <leo.singer@ligo.org>
-2008-04-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ Revert "Revert "audioresample: Add GAP flag support""
+ This reverts commit 35c76b3409dde7f2dcc8232388a47a1b99b661a7.
+ Conflicts:
+ gst/audioresample/gstaudioresample.c
+ gst/audioresample/gstaudioresample.h
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Filter cdda from the supported URI schemes. We can't support
- musicbrainz tags and everything else one expects from a cdda source
- with GIO. Fixes bug #526794.
+2010-12-16 10:26:43 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/pango/gsttextoverlay.c:
+ timeoverlay: add missing break
+ https://bugzilla.gnome.org/show_bug.cgi?id=637377
- * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_buffer_alloc):
- Fix calculation of 'expected size' for YV12 buffers.
- Be a little more verbose in the debug output for buffer-alloc'ed
- buffers which turn out to have the wrong size.
+2010-12-16 10:11:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Change classification to Filter/Converter/Video/Scaler
- * NEWS:
- * RELEASE:
- * gst-plugins-base.doap:
- Merge other changes from 0.10.19 release branch.
+2010-12-15 23:47:29 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstrtsp.def:
+ win32: update the def file with the new rtsp api
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init):
- * gst/playback/gstplayback.c: (plugin_init):
- * gst/volume/gstvolume.c: (plugin_init):
- Work around missing bits of thread-safety on older GLibs some
- more to avoid assertions when starting up multiple playbin
- objects concurrently (see #512382).
+2010-12-15 17:51:36 +0100 Andy Wingo <wingo@oblong.com>
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+ add gst_rtsp_url_decode_path_components
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ * gst-libs/gst/rtsp/gstrtspurl.c (gst_rtsp_url_decode_path_components):
+ New public function, returns a strv of uri-decoded path components.
+ * tests/check/Makefile.am:
+ * tests/check/libs/rtsp.c: Add tests.
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- Remove some more fields.
+2010-12-15 16:35:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * win32/common/libgstrtp.def:
+ win32: update defs file
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+2010-12-15 16:30:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Actually build dlls when cross-compiling with mingw32.
- Fixes bug #526247.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: relax arrangement for RTP bufferlists
+ Don't assume there are exactly 2 buffers but allow cases where the header and
+ payload are in 1 buffer or where the payload is in more buffers.
-2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-15 14:55:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
+ * common:
+ Automatic update of common submodule
+ From 20742ae to 169462a
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-15 12:58:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
- (msg_buffering), (connect_bus_signals), (main):
- Add statusbar.
- Add buffering support with feedback in the statusbar.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ basedepay: add support for buffer lists in the depayloader
+ Add support for buffer lists in the depayloader.
-2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-13 10:08:47 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
- * ext/ogg/gstoggmux.c:
- Fix sample pipeline description.
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/encoding/.gitignore:
+ * tests/examples/encoding/Makefile.am:
+ * tests/examples/encoding/encoding.c:
+ * tests/examples/encoding/gstcapslist.c:
+ * tests/examples/encoding/gstcapslist.h:
+ examples: encoding example
+ Along with gstcapslist
-2008-04-03 Stefan Kost <ensonic@users.sf.net>
+2010-08-13 17:36:38 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+ * configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
- Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
-
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-cdparanoia.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gdp.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playback.xml:
- * docs/plugins/inspect/plugin-queue2.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-uridecodebin.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml:
- Update introspection data.
-
- * ext/ogg/gstoggmux.c:
- Document oggmux.
-
- * gst/playback/gstdecodebin2.c:
- Don't use gtk-doc style comment start for private stuff, but make it
- formatted like this for consistency.
-
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (analyze_new_pad), (connect_pad), (expose_pad),
- (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
- (gst_decode_group_expose), (gst_decode_group_free),
- (do_async_start), (do_async_done), (gst_decode_bin_change_state):
- Remove fakesink hack, we can now implement this more elegantly.
- Added property to bypass typefinding.
- Removed underrun callback and demuxer pad probe, we now use the srcpad
- probe to expose groups.
- API::sink-caps property
-
- * gst/playback/gstplaybin2.c: (no_more_pads_cb):
- Guard against multiple emissions of the no_more_pads signal, which
- happens when we are dealing with chained oggs.
-
- * gst/playback/gsturidecodebin.c: (remove_decoders),
- (make_decoder), (type_found), (setup_streaming), (source_new_pad),
- (setup_source):
- For streams, use our own typefind element and plug our queue after it.
- We will need this to determine the type of buffering to use for the
- queue soon.
-
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
- Guard against over and underflows because of clock slaving.
- When we are using our own clock, still compensate for any calibrations
- that we might have done to our clock.
-
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/theora/theoradec.c: (theora_handle_type_packet),
- (theora_dec_chain):
- Don't try to do anything fancy with the return code from pushing an
- event, it does not have enough information to turn it into a
- GST_FLOW_ERROR.
-
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
- (gst_ogg_demux_chain_elem_pad):
- Add small debug line.
- Pass return code from the internal decoder instead of the too generic
- GST_FLOW_ERROR.
-
-2008-04-03 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/cdda/base64.c:
- * gst-libs/gst/cdda/base64.h:
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cddabasesrc_calculate_musicbrainz_discid):
- Use GLib's base64 implementation instead of our own.
-
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_read_chain):
- Refix oggdemux, we only have a problem if we failed to find a chain and
- we are not EOF.
-
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_read_chain):
- When we fail to find a BOS page and we and up with no chain, error out
- properly instead of segfaulting. Fixes #525665.
-
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
- The new-pad-group sequence is add-pads, no-more-pads, add-pads,
- no-more-pads...
+ * gst/encoding/.gitignore:
+ * gst/encoding/Makefile.am:
+ * gst/encoding/gstencode-marshal.list:
+ * gst/encoding/gstencodebin.c:
+ * gst/encoding/gstencodebin.h:
+ * gst/encoding/gstsmartencoder.c:
+ * gst/encoding/gstsmartencoder.h:
+ * gst/encoding/gststreamcombiner.c:
+ * gst/encoding/gststreamcombiner.h:
+ * gst/encoding/gststreamsplitter.c:
+ * gst/encoding/gststreamsplitter.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/encodebin.c:
+ gst: New encoding plugin
+ https://bugzilla.gnome.org/show_bug.cgi?id=627476
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-08-13 17:27:52 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
- * gst/playback/gstqueue2.c: (update_out_rates),
- (gst_queue_open_temp_location_file),
- (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
- (gst_queue_handle_src_query), (gst_queue_set_property):
- Update the estimated input data when we push out a buffer.
- Add some debug info about the temp file.
- Only forward src events when we are not using a temp file.
- Don't block the duration query, we need to find something better.
- Don't leak the temp filename.
+ * docs/design/Makefile.am:
+ * docs/design/design-encoding.txt:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/encoding-profile.c:
+ * gst-libs/gst/pbutils/encoding-profile.h:
+ * gst-libs/gst/pbutils/encoding-target.c:
+ * gst-libs/gst/pbutils/encoding-target.h:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/profile.c:
+ * win32/common/libgstpbutils.def:
+ pbutils: New Profile library
+ https://bugzilla.gnome.org/show_bug.cgi?id=627476
-2008-04-01 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-15 12:21:05 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
- Require GLib 2.12 and liboil 0.3.14.
-
- * gst/volume/gstvolume.c: (volume_process_double):
- Unconditionally use liboil 0.3.14 function.
-
-2008-03-31 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- ms-gsm can have arbitrarty sample rates. See #481354.
-
-2008-03-28 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- MP4S is generic MPEG-4, not a microsoft variant.
+ configure: use the -Bsymbolic-functions linker flag if supported
+ This feature turns intra library calls into direct function calls and thus makes
+ them a little faster. The downside is that this causes problems for e.g.
+ LD_PRELOAD based tools. Thus add a configure option to turn it off.
-2008-03-27 Michael Smith <msmith@fluendo.com>
+2010-12-14 00:16:13 -0800 David Schleef <ds@schleef.org>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Check the body CRC (if set) when depayloading.
- Fixes #522401.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add check for yuv4mpeg
-2008-03-24 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-13 18:05:41 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
- Fix Since: version for new property.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: spell out two more container formats
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-13 16:20:23 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
- (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
- Don't error when poll_wait returns EAGAIN.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstqueue2.c: (gst_queue_is_filled):
- The queue is never filled when there are no buffers in the queue at all.
- Fixes #523993.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (init_group), (free_group), (gst_play_bin_init),
- (gst_play_bin_finalize), (gst_play_bin_set_uri),
- (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
- (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
- (gst_play_bin_set_current_video_stream),
- (gst_play_bin_set_current_audio_stream),
- (gst_play_bin_set_current_text_stream),
- (gst_play_bin_set_encoding), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
- (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
- (activate_group), (deactivate_group), (setup_next_source),
- (save_current_group), (gst_play_bin_change_state):
- Update some docs.
- Add new locks and conds to protect pipeline creation and group
- switching.
- Implement the sub-uri property.
- Keep track of pending uridecodebin creation and configure the output
- pipeline after all streams are configured.
- Propagate subtitle encoding to the uridecodebins.
- Implement getting the video/audio/visualisation elements.
- Use input-selector for stream switching.
- If we are asked to do visualisation, prefer to autoplug raw sinks
- instead of sinks that accept encoded data.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
- (gst_play_sink_init), (gst_play_sink_dispose),
- (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
- (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
- (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
- (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
- (gst_play_sink_set_volume), (gst_play_sink_get_volume),
- (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
- (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
- (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Add methods to get audio/video/vis elements.
- Add methods to set the font description for the overlay.
- Remove properties, we're using this element with its methods only.
- Add support for subtitles.
- Rearrange the locking a bit to not use the object lock for protecting
- the pipeline construction.
- Try to use the volume and mute property on the sink when its available.
- Implement the mute option with volume when the sink does not have a mute
- property.
- Only add volume element when the sink has no volume property.
- Only do visualisations with raw audio pads.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
- (gst_text_overlay_init), (gst_text_overlay_set_property),
- (gst_text_overlay_get_property), (gst_text_overlay_src_event),
- (gst_text_overlay_text_event), (gst_text_overlay_video_event),
- (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
- (gst_text_overlay_change_state):
- * ext/pango/gsttextoverlay.h:
- Add property to configure waiting for text on the textpad or not, with
- the default behaviour being the old one (always wait for text before
- rendering the video). This default behaviour is usually not the best one
- because the text stream can very sparse and could require queueing a lot
- of video.
- Fix the flushing and EOS handing so that we don't mix up their meaning.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ * tools/gst-discoverer.c:
+ * win32/common/libgstpbutils.def:
+ discoverer: query seekability
+ Besides the duration we can also query the seekability of a stream. Use the new
+ API in the gst-discoverer tool.
+ API: gst_discoverer_info_get_seekable
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-13 16:23:04 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gsturidecodebin.c:
- (gst_uri_decode_bin_autoplug_factories),
- (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
- (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
- (gst_uri_decode_bin_set_property),
- (gst_uri_decode_bin_get_property), (no_more_pads_full),
- (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
- (proxy_autoplug_factories_signal), (make_decoder),
- (source_new_pad), (setup_source):
- Add a readonly source property and notify.
- Add new lock for protecting the construction of the pipeline.
- Keep track of the decodebins we plugged.
- Correctly proxy the autoplug signal so that it actually continues.
- Proxy subtitle-encoding to the decodebins.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
- (text_toggle_cb), (update_streams), (main):
- Rearrange some buttons in playbin2 and make some other boxes insensitive
- when needed.
- Add language codes to subtitle selection boxes when we gind the right
- tags for the streams.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
- (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
- (gst_decode_bin_set_subs_encoding),
- (gst_decode_bin_get_subs_encoding),
- (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
- (deactivate_free_recursive):
- Protect caps property with the object lock.
- Protect encoding property with the object lock.
- Keep list of elements we added that have the subtitle-encoding property.
- Distribute the subtitle-encoding to all of the elements when it
- changes.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
- Small debug improvement.
+ * common:
+ Automatic update of common submodule
+ From 011bcc8 to 20742ae
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Fix bug in determining the sample start/stop position, we want to base
- this decision on the fact that we are going forwards or backwards, not
- slower or faster. This fixes some ugly resync warnings when playing at
- very slow speeds.
+2010-12-13 13:04:40 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/check/elements/audioresample.c:
+ tests: audioresample: adjust unit test to relaxed discont checking
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Correctly set the supported URI schemes and don't leave
- some schemes in the middle or at the start at NULL.
+2010-12-13 12:34:58 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-23 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/Makefile.am:
+ * docs/design/Makefile.am:
+ make: move the design doc also on the Makefile.am level (for dist)
- * tests/check/elements/gdpdepay.c:
- Make test compile without unused function/variable warnings on PPC.
-
-2008-03-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-13 10:05:00 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_class_init):
- * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_class_init):
- * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
- * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
- * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
- * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
- * ext/pango/gsttextrender.c: (gst_text_render_class_init):
- * ext/theora/theoradec.c: (gst_theora_dec_class_init):
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
- * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_class_init):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init):
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_class_init):
- * gst-libs/gst/interfaces/mixertrack.c:
- (gst_mixer_track_class_init):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init):
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init):
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
- (gst_audioresample_class_init):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_class_init):
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (preroll_unlinked):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
- * gst/playback/gstqueue2.c: (gst_queue_class_init):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
- * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
- (gst_stream_selector_class_init):
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
- * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_class_init):
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
- * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
- static strings (i.e. all). This gives us less memory usage,
- fewer allocations and thus less memory defragmentation. Depend
- on core CVS for this. Fixes bug #523806.
-
-2008-03-22 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Filter http and https protocols. GIO/GVfs handles them but it's
- impossible to implement iradio/icecast with it. Better use
- souphttpsrc or something else for this.
-
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
- If getting the file informations by a query fails try it with the
- seek-to-end trick too.
-
-2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/volume/gstvolume.c: (gst_volume_interface_supported),
- (gst_volume_base_init), (gst_volume_class_init),
- (volume_process_double), (volume_process_float),
- (volume_transform_ip), (plugin_init):
- memset buffers to zero if we get a GAP buffer. We usually see a
- buffer as one unit so let's handle it as one and don't care about
- volume changes while processing one buffer.
- Also clean up some stuff a bit.
-
-2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_create_silence_buffer),
- (gst_audio_convert_transform):
- Make audioconvert GAP-aware by outputting silence buffers when the
- input has the GAP flag set. This is up to 8x faster.
- Based on a patch by Stefan Kost. Fixes bug #517813.
-
-2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/volume/gstvolume.c: (volume_process_double):
- Use oil_scalarmultiply_f64_ns() for double processing when it's
- available at compile time.
-
-2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
+ audioresample: relax discont checking slightly
- * configure.ac:
- Fix lrint/lrintf checks to actually work. These functions are
- in libm on Linux at least so try to link to it.
+2010-12-13 09:56:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-21 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: provide as much valid output ts and offset as valid input
+ ... by independently tracking time and offset, rather than having no offset
+ leading to no output ts.
- * configure.ac:
- Back to development - 0.10.18.1
+2010-12-13 10:41:24 +0200 Stefan Kost <ensonic@users.sf.net>
-=== release 0.10.18 ===
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinders: name "aac" typefinder "audio/aac"
+ This is in sync how we call the others.
-2008-03-20 Jan Schmidt <jan.schmidt@sun.com>
+2010-12-13 09:58:53 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- releasing 0.10.18, "I will follow"
+ * docs/design-audiosinks.txt:
+ * docs/design/design-audiosinks.txt:
+ docs: move design doc to design folder
-2008-03-18 Jan Schmidt <jan.schmidt@sun.com>
+2010-12-11 19:33:33 +0200 Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
- * configure.ac:
- * win32/common/config.h:
- 0.10.17.4 pre-release
+ * gst/videotestsrc/generate_sine_table.c:
+ videotestsrc: Add a missing return statement
-2008-03-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-11 17:18:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
- Use GST_STR_NULL when trying to print strings that could be NULL because
- this might crash on some platforms. See #520808.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Deprecate new-decoded-pad and removed-decoded-pad signals
+ They're really the same as pad-added and pad-removed from GstElement
+ and it doesn't make sense to have two signals for the same thing.
-2008-03-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-11 17:14:36 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Emit "remove-decoded-pad" signal when pads are removed from decodebin2
+ Fixes bug #636198.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
- (read_line), (gst_rtsp_connection_read_internal):
- Generic Windows fixes that makes libgstrtsp work on Windows when
- coupled with the new GstPoll API. See #520808.
+2010-12-10 18:57:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-03-17 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: unset flushing flag when starting
+ When we start again after being stopped, clear the flushing flag or else
+ it will always be TRUE.
+ Fixes #636769
- Patch by: Milosz Derezynski <internalerror at gmail dot com>
+2010-12-09 16:57:35 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
- If seeking to a new position succeeds don't simply return from
- create() without creating a buffer. Do this only in the case
- seeking to the new position fails. Fixes bug #523054.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add/Fix some media descriptions
+ Fixes #623413
-2008-03-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-09 08:40:25 +0100 Gavin Stark <g.stark@realdigitalmedia.com>
- * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
- (gst_video_format_from_rgba32_masks):
- Fix gst_video_format_parse_caps() for RGB caps with alpha channel
- (#522635).
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Use gst_caps_can_intersect() instead of gst_caps_intersect()
+ Fixes a memory leak and bug #636827.
- * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
- Add unit test for the RGB caps parsing and creation, checking for
- internal consistency of the new API and consistency of the API with
- the old GST_VIDEO_CAPS_* defines.
+2010-12-08 12:55:24 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-14 David Schleef <ds@schleef.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: improve iso media typefinding
+ ... by also considering compatible brands rather than only aiming at major brand
+ (of which there are a seemingly ever expanding great many).
- * gst/videotestsrc/videotestsrc.c: Oops, revert last change
- because -base is in freeze.
+2010-12-08 12:28:32 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-14 David Schleef <ds@schleef.org>
+ * tests/check/libs/pbutils.c:
+ tests: remove superflous ';' and reindent
- Patch by: William M. Brack
+2010-12-08 12:09:45 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ docs: fix wrong use of Since: keyword
-2008-03-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-07 20:28:37 +0200 René Stadler <rene.stadler@nokia.com>
- * gst/playback/gststreamselector.c: (gst_selector_pad_event),
- (gst_selector_pad_chain):
- * gst/playback/gststreamselector.h:
- Revert change that caused regression until a real fix is found.
- Fixes #522203.
+ * tests/check/gst/typefindfunctions.c:
+ tests: add AC-3, E-AC-3 typefind tests
-2008-03-12 Michael Smith <msmith@fluendo.com>
+2010-12-03 17:33:40 +0200 René Stadler <rene.stadler@nokia.com>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
- * gst-libs/gst/audio/gstringbuffer.h:
- Rename recently added buffer types to make more sense.
- * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
- (gst_alsasink_write):
- Adapt for above API changes.
- Fixes bug #520523.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: ignore AC-3 BSIDs 9, 10 and >16
+ These are reserved for future extensions which will not be backwards
+ compatible to E-AC-3.
-2008-03-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-03 16:54:21 +0200 René Stadler <rene.stadler@nokia.com>
- * win32/common/libgstnetbuffer.def:
- Add new symbol gst_netaddress_equal. Fixes bug #521743.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: accept consecutive AC-3 frames of different sizes
+ This is perfectly valid and occurs in particular when there are
+ (in)dependent substreams present.
-2008-03-11 Jan Schmidt <jan.schmidt@sun.com>
+2010-12-03 16:22:32 +0200 René Stadler <rene.stadler@nokia.com>
- * configure.ac:
- * win32/common/config.h:
- 0.10.17.3 pre-release
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: remove useless masking in (E-)AC-3 typefinders
-2008-03-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-03 16:14:15 +0200 René Stadler <rene.stadler@nokia.com>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Fix duration when no clock was provided. Fixes #520300.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: stop scanning after suggesting E-AC-3 caps
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-12-03 18:08:58 +0200 René Stadler <rene.stadler@nokia.com>
- Patch by: Olivier Crete <tester at tester ca>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix E-AC-3 frame size parsing
+ Frame size is given in words; it is already multiplied by two where
+ needed, so the left shift is superfluous. This extra multiplication
+ caused the code to inspect the third packet instead of the second,
+ which would fail for files where the second packet has a size
+ different from the first.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- Add trivial function to compare GstNetAddress. See #520626.
- API: GstNetBuffer::gst_netaddress_equal
+2010-12-07 17:35:14 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: Move around the typedefs to make GIR happy
+ Otherwise it will generate they symbols as _GstRTSP* (with the leading
+ underscore).
+
+2010-12-04 14:48:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ * tests/examples/app/appsrc-stream.c:
+ * tests/examples/app/appsrc-stream2.c:
+ tests: use GLib 2.22 API unconditionally
+
+2010-12-04 14:45:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/tag/lang.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ * gst-libs/gst/video/convertframe.c:
+ libs: use GLib 2.22 API unconditionally
+
+2010-12-03 17:41:18 +0100 Benjamin Gaignard <benjamin.gaignard@stericsson.com>
+
+ * Android.mk:
+ * android/NOTICE:
+ * android/alsa.mk:
+ * android/app.mk:
+ * android/app_plugin.mk:
+ * android/audio.mk:
+ * android/audioconvert.mk:
+ * android/decodebin.mk:
+ * android/decodebin2.mk:
+ * android/gdp.mk:
+ * android/gst-libs/gst/app/gstapp-marshal.c:
+ * android/gst-libs/gst/app/gstapp-marshal.h:
+ * android/gst-libs/gst/audio/audio-enumtypes.c:
+ * android/gst-libs/gst/audio/audio-enumtypes.h:
+ * android/gst-libs/gst/interfaces/interfaces-enumtypes.c:
+ * android/gst-libs/gst/interfaces/interfaces-enumtypes.h:
+ * android/gst-libs/gst/interfaces/interfaces-marshal.c:
+ * android/gst-libs/gst/interfaces/interfaces-marshal.h:
+ * android/gst-libs/gst/pbutils/pbutils-enumtypes.c:
+ * android/gst-libs/gst/pbutils/pbutils-enumtypes.h:
+ * android/gst-libs/gst/rtsp/gstrtsp-enumtypes.c:
+ * android/gst-libs/gst/rtsp/gstrtsp-enumtypes.h:
+ * android/gst-libs/gst/rtsp/gstrtsp-marshal.c:
+ * android/gst-libs/gst/rtsp/gstrtsp-marshal.h:
+ * android/gst-libs/gst/video/video-enumtypes.c:
+ * android/gst-libs/gst/video/video-enumtypes.h:
+ * android/gst/playback/gstplay-marshal.c:
+ * android/gst/playback/gstplay-marshal.h:
+ * android/gst/tcp/gsttcp-enumtypes.c:
+ * android/gst/tcp/gsttcp-enumtypes.h:
+ * android/gst/tcp/gsttcp-marshal.c:
+ * android/gst/tcp/gsttcp-marshal.h:
+ * android/interfaces.mk:
+ * android/netbuffer.mk:
+ * android/pbutils.mk:
+ * android/playbin.mk:
+ * android/queue2.mk:
+ * android/riff.mk:
+ * android/rtp.mk:
+ * android/rtsp.mk:
+ * android/sdp.mk:
+ * android/tag.mk:
+ * android/tcp.mk:
+ * android/typefindfunctions.mk:
+ * android/video.mk:
+ Add build system for Android
+
+2010-12-03 15:46:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Update mode property docs, it's deprecated now.
+ * win32/common/libgstvideo.def:
+ defs: add new symbol
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-27 13:49:41 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_create):
- * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
- (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
- Remove GstPollMode from gstpoll constructor.
+ * ext/ogg/gstoggstream.c:
+ oggstream: additional tag extraction
+ ... supporting theora, flac, speex, celt.
+ Fixes #629349.
-2008-03-04 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-27 12:08:25 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- * win32/common/config.h:
- 0.10.17.2 pre-release
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggstream: use separate tag extraction vfunction
-2008-03-03 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-27 11:58:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/Makefile.am:
- GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
- them twice
+ * ext/ogg/gstoggstream.c:
+ oggstream: refactor vorbis comment tag extraction
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Add new API to the defs
+2010-10-27 11:16:15 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: plug some oggstream leaks
- Patch by: Mersad Jelacic <mersad at axis dot com>
+2010-10-27 10:59:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
- possible to specify the sample size in bits. (#509637)
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggstream: streamline tag extraction and prevent some leaks
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-27 10:58:16 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/libs/mixer.c:
- Add a few simple checks for the new message types.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: send stream tags after newsegment and global tags
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-14 23:08:51 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
- (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
- (gst_mixer_message_get_type),
- (gst_mixer_message_parse_option_changed),
- (gst_mixer_message_parse_options_list_changed):
- * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
- (GST_MIXER_MESSAGE_OPTION_CHANGED),
- (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
- (GST_MIXER_MESSAGE_MIXER_CHANGED):
- API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
- and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: perform more (vorbis comment header) tag extractions
+ In particular, move comment header parsing to gstoggstrem.c.
+ Thanks to Felipe Contreras.
+ Fixes #629349 (partially).
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-27 10:20:15 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
- (gst_mixer_options_get_values):
- * gst-libs/gst/interfaces/mixeroptions.h:
- (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
- (_GstMixerOptions), (_GstMixerOptionsClass):
- API: add GstMixerOptions::get_values vfunc (#519906)
+ * gst-libs/gst/riff/riff-ids.h:
+ riff: document omitted field in _gst_riff_strf_auds
+ (aka WAVEFORMATEX)
-2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+2010-10-10 17:15:53 -0700 David Schleef <ds@schleef.org>
- * configure.ac:
- Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
- plug-ins are included/excluded. (#498222)
+ * ext/ogg/gstoggstream.c:
+ oggstream: fix incorrect warning on skeleton headers
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-11-20 19:02:50 -0800 David Schleef <ds@schleef.org>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for IMelody files, using audio/x-imelody.
- See bug #519516.
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggparse: Set DELTA_UNIT on buffers
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-12-03 00:01:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
- * ext/alsa/gstalsasink.c: (set_hwparams):
- * ext/alsa/gstalsasrc.c: (set_hwparams):
- * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
- * ext/ogg/gstoggmux.h:
- * ext/ogg/gstogmparse.c:
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_uri_sink_message_new),
- (gst_missing_element_message_new),
- (gst_missing_decoder_message_new),
- (gst_missing_encoder_message_new):
- * gst-libs/gst/rtp/gstbasertppayload.c:
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_bye_get_reason):
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/ffmpegcolorspace/imgconvert.c:
- * gst/playback/test.c: (gen_video_element), (gen_audio_element):
- * gst/typefind/gsttypefindfunctions.c:
- * gst/videoscale/vs_4tap.c:
- * gst/videoscale/vs_4tap.h:
- * sys/v4l/gstv4lelement.c:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
- * sys/v4l/v4l_calls.c:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
- (gst_v4lsrc_try_capture):
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new):
- * tests/check/elements/audioconvert.c:
- * tests/check/elements/audioresample.c:
- (fail_unless_perfect_stream):
- * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
- * tests/check/elements/decodebin.c:
- * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
- (setup_gdpdepay_streamheader):
- * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
- (setup_gdppay_streamheader):
- * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
- * tests/check/elements/multifdsink.c: (setup_multifdsink):
- * tests/check/elements/textoverlay.c:
- * tests/check/elements/videorate.c: (setup_videorate):
- * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
- * tests/check/elements/volume.c: (setup_volume):
- * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
- * tests/check/elements/vorbistag.c:
- * tests/check/generic/clock-selection.c:
- * tests/check/generic/states.c: (setup), (teardown):
- * tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
- * tests/check/pipelines/gio.c:
- * tests/check/pipelines/oggmux.c:
- * tests/check/pipelines/simple-launch-lines.c:
- (simple_launch_lines_suite):
- * tests/check/pipelines/streamheader.c:
- * tests/check/pipelines/theoraenc.c:
- * tests/check/pipelines/vorbisdec.c:
- * tests/check/pipelines/vorbisenc.c:
- * tests/examples/seek/scrubby.c:
- * tests/examples/seek/seek.c: (query_positions_elems),
- (query_positions_pads):
- * tests/icles/stress-xoverlay.c: (myclock):
- Correct all relevant warnings found by the sparse semantic code
- analyzer. This include marking several symbols static, using
- NULL instead of 0 for pointers and using "foo (void)" instead
- of "foo ()" for declarations.
-
- * win32/common/libgstrtp.def:
- Add gst_rtp_buffer_set_extension_data to the symbol definition file.
+ tests: fix video library unit test and skip non-working YUV9/YVU9 parts for now
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-12-02 23:49:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: José Alburquerque <jaalburqu svn gnome org>
+ * gst-libs/gst/video/video.c:
+ video: add missing break statement for the GST_VIDEO_FORMAT_RGB8_PALETTED case
- * gst/playback/gstplaybin2.c:
- Make the function signature of the _get_*_tags() functions match
- the signature of the vfuncs they implement, ie. return a
- GstTagList rather than a GstStructure, which is more correct,
- even if one is typedef'ed to the other (#518940).
+2010-11-15 22:02:07 +0200 Evan Broder <evan@ebroder.net>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * tools/gst-visualise-m.m:
+ gst-visualise: trim unused perl dependency
+ Remove an unused perl module. Fixes #634522.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Don't include unix headers unconditionally (fixes #518037).
+2010-11-01 23:07:12 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: add some logging for failure case
- * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
- (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
- (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
- (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
- (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
- (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
- (gst_video_format_is_packed), (video_format_is_packed):
- Add unit test that makes sure that the strides, offsets and
- sizes returned for the various YUV formats by the new video API
- match the old reference implementation in videotestsrc.
+2010-11-01 23:06:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstinputselector.c:
+ inputselector: log times in human readable form
- * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
- (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
- (gst_video_format_is_rgb), (gst_video_format_is_yuv),
- (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
- (gst_video_format_get_pixel_stride),
- (gst_video_format_get_component_width),
- (gst_video_format_get_component_height),
- (gst_video_format_get_component_offset), (gst_video_format_get_size):
- * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
- (GST_VIDEO_FORMAT_Y42B):
- API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
+2010-11-01 22:44:16 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstinputselector.c:
+ inputselector: more G_PARAM_STATIC_STRINGS use
- * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
- YV12 is I420 with swapped components 1 and 2, so the offset of
- component 1 for I420 should be the offset for component 2 for YV12
- and vice versa.
+2010-11-01 22:42:23 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-29 Rene Stadler <mail@renestadler.de>
+ * gst/playback/gstinputselector.c:
+ inputselector: move reoccuring logs to LOG and remove a double info
+ Less debug spew in DEBUG category. No need to log pad again if we use
+ GST_LOG_OBJECT(pad,...).
- * sys/v4l/gstv4lelement.c:
- Add missing semicolon to fix indentation.
+2010-12-02 19:11:37 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-02-29 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/rtsp/Makefile.am:
+ libgstrtsp: Fix typo in .pc to use for GIR
- * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
- (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to detect
- if we can do SPDIF output.
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
- (gst_alsasink_prepare), (gst_alsasink_close), (gst_alsasink_write):
- * ext/alsa/gstalsasink.h: Initial support for SPDIF.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
- * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer types
- to support AC3, EC3 and IEC958 buffers.
+2010-12-02 15:16:25 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-02-29 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ docs: Add a whole bunch of symbols that were unused to the proper sections
- * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
- (gst_mixer_message_parse_mute_toggled),
- (gst_mixer_message_parse_record_toggled),
- (gst_mixer_message_parse_volume_changed),
- (gst_mixer_message_parse_option_changed):
- De-cruft and fix message type assertions (NULL is not a really
- valid mixer message type string).
+2010-11-10 11:02:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-29 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: only parse TTL for IP4 addresses
+ Only IP4 addresses can have a TTL in the address.
- * ext/libvisual/visual.c: (gst_vis_src_negotiate):
- When negotiating, actually start from a format that we can support
- instead of from the too generic template.
+2010-11-10 10:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-29 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ * win32/common/libgstsdp.def:
+ sdp: add method to check for multicast addresses
+ Expose a previously internal method to check for multicast addresses.
+ See #634093
- * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
- Enable vis setting.
+2010-11-03 11:13:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaysink.c: (gst_play_sink_init),
- (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
- (gen_vis_chain):
- Implement vis switching while playing.
+ * gst-libs/gst/pbutils/gstpluginsbaseversion.h.in:
+ pbutils: Take nano version into account in GST_CHECK_PLUGINS_BASE_VERSION()
+ If the nano is > 0 the current version should be handled the same as
+ micro + 1.
-2008-02-28 David Schleef <ds@schleef.org>
+2010-11-03 09:51:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add YUV9, YVU9 and IYU1 video formats
+ API: GST_VIDEO_FORMAT_YUV9: planar 4:1:0 YUV
+ API: GST_VIDEO_FORMAT_YVU9: planar 4:1:0 YUV (chroma planes swapped)
+ API: GST_VIDEO_FORMAT_IYU1: packed 4:1:1 YUV (Cr-Y0-Y1-Cb-Y2-Y3)
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-11-02 11:57:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add 8-bit paletted RGB
+ API: Add GST_VIDEO_FORMAT_RGB8_PALETTED
+ API: Add GST_VIDEO_CAPS_RGB8_PALETTED
+ API: Add gst_video_parse_caps_palette()
- * gst/tcp/Makefile.am:
- * gst/tcp/fdsetstress.c:
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstfdset.h:
- Removed fdset and stress test, they are now known as GstPoll in
- core.
-
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
- (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_queue_buffer),
- (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
- (gst_multi_fd_sink_stop):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
- (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
- (gst_tcp_gdp_read_caps):
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
- (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
- (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
- (gst_tcp_client_src_create), (gst_tcp_client_src_start),
- (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
- (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
- (gst_tcp_server_src_create), (gst_tcp_server_src_start),
- (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
- * gst/tcp/gsttcpserversrc.h:
- Port to GstPoll. See #505417.
+2010-10-31 19:17:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfssrc: Remove dead assignment
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2010-10-31 19:14:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
- (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
- (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
- (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
- (gst_rtsp_connection_flush):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Use GstPoll for the rtsp connection. See #505417.
+ * gst/tcp/gsttcp.c:
+ tcp: Remove dead assignment
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-31 19:11:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
- (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
- Add combo box for visualisations, populate it with a factory list
- of all visualisation plugins, configure vis plugin instance in
- playbin2.
+ * gst/playback/gstplaysink.c:
+ playsink: gen_video_chain() always returns a bin, no need to check for that
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-31 19:08:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/rtp.c: (GST_START_TEST):
- Add check for RTP buffer defaults, padding and marker bit API.
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: If we get EOS for an unknown stream just do nothing
+ instead of dereferencing NULL pointers. This can happen if the stream
+ was just removed from the streamsynchronizer in a bad time.
-2008-02-27 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-31 19:06:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/cdda/sha1.c: (sha_transform):
- Use memcpy() instead of upcasting a byte array to long *. This
- fixes an unaligned memory access, resulting in SIGBUS on IA64.
- This should be ported to GCheckSum once we can use GLib 2.16.
- Partially fixes bug #500833.
+ * gst/playback/gstplaysink.c:
+ playsink: gen_video_deinterlace_chain() always returns a bin, no need to check that
-2008-02-27 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-31 19:01:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
- Push tag event after the newsegment event. Log the pointer of
- the buffer we're actually going to push rather than the buffer
- we're feeding to _make_metadata_writable().
+ * sys/v4l/v4l_calls.c:
+ v4l: If no video tuner is the requested one don't read unitialized data
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-25 14:13:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Comment smoke typefinder for now. The smokedec plugin needs one
- frame per buffer but we have no parser yet, thus it simply crashes
- in most situations.
+ * sys/ximage/ximagesink.c:
+ ximagesink: Add docs for the new property
+ Including Since markers
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-25 14:11:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for the smoke video codec. Copied from the jpeg plugin.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Add docs for the new property
+ Including Since markers
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-25 14:09:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
- (plugin_init):
- Add midi typefinder, copied from the timidity plugin.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Use PROP_ instead of ARG_ for the property enums
-2008-02-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-25 14:09:20 +0100 Andrea Sebastianutti <andreailseba@virgilio.it>
- Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Add read-only properties window-width and window-height
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * tests/check/elements/subparse.c: (test_microdvd_with_italics),
- (subparse_suite):
- Forward slashes at the beginning and end of a line also signify
- italics (Fixes: #518162).
+2010-10-25 14:08:43 +0100 Andrea Sebastianutti <andreailseba@virgilio.it>
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+ * sys/ximage/ximagesink.c:
+ ximagsink: Add read-only properties window-width and window-height
- * tests/check/gst-plugins-base.supp:
- Add a suppression for a cached value in GIO that wasn't moved
- while moving gio from -bad to -base.
+2010-10-17 14:26:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/video/video.c:
+ video: Return correct component width/height for A420
- Patch by: Brian Cameron <brian dot cameron at sun dot com>
+2010-12-02 00:15:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- Don't hardcode -Wall and -Werror for configure checks, this fails
- with non-GCC compilers. Fixes bug #517991.
-
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
-
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audioconvert/gstaudioquantize.c:
- * gst/audioconvert/gstaudioquantize.h:
- Make audioconvert gap aware. If noiseshaping is used, wait for
- noiseshaping to converge before marking as GAP. Fixes #517813.
+ Bump GLib requirement to >= 2.22
+ See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
+2010-12-02 00:12:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
-
-2008-02-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/gnomevfs/gstgnomevfssink.c:
- (gst_gnome_vfs_sink_handle_event):
- Return FALSE when seeking for a new segment fails instead
- of silently ignoring the failure and appending every buffer
- that comes for the new segment.
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development
-2008-02-20 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.31 ===
- * gst/playback/gstplaysink.c: (find_property),
- (gst_play_sink_find_property), (gen_video_chain),
- (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
- Recursively search the sink element for a last-frame property so that we
- can also find the property in autovideosink and friends that don't
- always proxy the internal sink properties.
+2010-11-30 19:25:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.31
- * gst-libs/gst/audio/multichannel.c:
- (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
- (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
- (gst_audio_set_structure_channel_positions_list),
- (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
- (gst_audio_fixate_channel_positions):
- Fix confusing terminology in docs and code: structure fields are
- 'fields' and not 'properties'.
+2010-11-24 17:34:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: disconnect signal handlers before disposing
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions), (add_list_to_struct):
- Give more useful warning messages if one of the channel
- layout enums passed to us is invalid and if the "channels"
- field in the caps has a GType we don't expect.
+2010-11-22 00:54:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ docs: improve decodebin2 docs a little
+ Mention that new pads may be created even after no-more-pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=634584
- * gst-libs/gst/audio/multichannel.c:
- Fix typo in docs blurb.
+2010-11-20 15:45:49 -0800 Evan Nemerson <evan@coeus-group.com>
-2008-02-19 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Add information on exported packages to GIRs
+ https://bugzilla.gnome.org/show_bug.cgi?id=635392
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2010-11-18 04:51:56 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
- typefind lookup to fix typefinding on HD clips.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ discoverer: Minor documentation fix
+ docs: Minor discoverer documentation fix
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-11-18 00:36:14 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstscreenshot.c:
- * gst/playback/gstscreenshot.h:
- Fix up copyright (I rewrote the GStreamer-0.10 code for
- this from scratch back in the days).
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.30.5 pre-release
-2008-02-19 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-11-18 00:35:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
- (create_element), (gst_play_frame_conv_convert):
- * gst/playback/gstscreenshot.h:
- Add screenshot conversion code from totem.
+ * po/bg.po:
+ * po/ca.po:
+ * po/es.po:
+ * po/hu.po:
+ * po/sk.po:
+ * po/tr.po:
+ po: update translations
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
- (gst_play_bin_class_init), (gst_play_bin_convert_frame),
- (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
- Implement frame property to get a color-unconverted snapshot.
- Implement convert-frame action signal to get a converted snapshot image.
- Configure connection speed in uridecodebin.
- Document some more properties.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
- (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_get_last_frame):
- * gst/playback/gstplaysink.h:
- Use last-buffer property of the video sink to get a video snapshot.
+2010-11-18 00:33:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (shot_cb), (main):
- Add snapshot button for playbin2 and use the frame property to save the
- frame as a png in the current directory.
+ * gst/playback/gstdecodebin.c:
+ decodebin: fix one more pad template ref leak
-2008-02-19 Sebastian Dröge <slomo@circular-chaos.org>
+2010-11-17 10:14:59 +0200 Harri Mähönen <harri.mahonen@gmail.com>
- Patch by: Josep Torra Valles <josep at fluendo dot com>
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ *decodebin*: don't leak pad templates set on ghostpads
+ https://bugzilla.gnome.org/show_bug.cgi?id=635067
- * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
- (plugin_init):
- Add typefinding support for h264 elementary streams.
- Fixes bug #517420.
+2010-11-17 01:01:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-18 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gststreamsynchronizer.c:
+ playbin2: disable streamsynchronizer magic for this release
+ Some things aren't quite right yet and cause problems (0-sized buffers
+ with PREROLL flag set cause crashes in elements that don't expect those;
+ getting pipeline back to preroll/playing again when audio/video streams
+ have different lengths and a seek past the end of one of the stream
+ happens doesn't always work, etc.). Needs further investigation in the
+ next cycle.
+ https://bugzilla.gnome.org/show_bug.cgi?id=633700
+ https://bugzilla.gnome.org/show_bug.cgi?id=634699
- * configure.ac:
- Require CVS of core for new API in collectpads.
+2010-11-08 09:27:52 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/adder/gstadder.c:
- Use new API to make adder sparse stream aware.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Fix a gtk-doc gobject-introspection annotation
+ gst_discoverer_discover_uri() expects the caller to unref the returned
+ GstDiscovererInfo object. The corresponding gtk-doc annotation was not
+ updated to reflect this.
-2008-02-18 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
- (no_more_pads_cb):
- Get the object data correct so that we can remove our channels
- correctly.
-
- * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
- (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_request_pad):
- Add option to disable async behaviour in the sinks when possible. This
- makes it possible to avoid an audio queue when dealing with
- visualisations.
- Add option to add a queue for the audio path.
-
- * tests/examples/seek/seek.c: (clear_streams), (update_streams),
- (main):
- Disable the vis checkbox to match the defaults of playbin2.
- Only get the stream info when we need to.
-
-2008-02-17 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
- (gst_gio_base_sink_set_stream):
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
- (gst_gio_base_src_set_stream):
- * ext/gio/gstgiosink.c: (gst_gio_sink_start):
- * ext/gio/gstgiosrc.c: (gst_gio_src_start):
- Don't use async operations as they require a running main loop.
- This makes us block again when closing streams and unable
- to mount the enclosing volume of an URI if it isn't yet.
-
-2008-02-15 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
- (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
- (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_request_pad):
- Move tee in front of the audio and vis pipelines.
- Add queue for audio for now.
- Add visualisation support.
-
- * tests/examples/seek/seek.c: (main):
- Visualisation is by default disabled.
-
-2008-02-15 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/gio/gstgiobasesink.c: (close_stream_cb):
- * ext/gio/gstgiobasesrc.c: (close_stream_cb):
- Improve debugging a bit.
-
- * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
- * ext/gio/gstgiosink.h:
- * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
- * ext/gio/gstgiosrc.h:
- Try to mount the enclosing volume of a GFile if it isn't mounted
- yet. This requires us to wait for an async operation to finish, done
- with an nested GMainLoop. Authentication is not supported yet, will
- come later.
-
-2008-02-14 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
- * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
- (gst_play_sink_get_mute), (gen_audio_chain):
- * gst/playback/gstplaysink.h:
- Add mute property.
+2010-11-08 09:26:27 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/playback/gststreamselector.c: (gst_selector_pad_event),
- (gst_selector_pad_chain):
- * gst/playback/gststreamselector.h:
- Make sure we forward the event only once.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * tools/gst-discoverer.c:
+ discoverer: Fix argument type to _container_info_get_streams()
+ No reason for gst_discoverer_container_info_get_streams() to not take a
+ GstDiscovererContainerInfo as its argument.
- * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
- Add and implement the mute button for playbin2.
+2010-11-05 20:47:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ configure: add --with-gtk option and default to Gtk+ 2.0 while the 3.0 API is still in flux
+ https://bugzilla.gnome.org/show_bug.cgi?id=634014
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2010-11-03 10:35:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
- Add some more debug info.
- Make sure we never return a negative delay. Fixes #516246.
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix IYU1 support
+ Fix conversions to IYU1, they allocated infinite amounts of memory before
+ because no conversion to IYU1 was actually implemented and it was running
+ into an infinite loop trying to find suitable intermediate formats.
+ Also fix the stride and sizes used for IYU1.
-2008-02-12 Tim-Philipp Müller <tim at centricular dot net>
+2010-11-02 12:29:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
- Revert patch that makes the sink hold the object lock when
- calling snd_pcm_delay(), since it breaks playback for me.
+ * tests/check/libs/rtp.c:
+ tests: fix invalid free and buffer list leak in rtp library unit test
-2008-02-12 Julien Moutte <julien@fluendo.com>
+2010-11-02 12:03:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
- some seek flags when changing rate.
+ * tests/check/libs/tag.c:
+ tests: fix leak in tag library unit test
-2008-02-12 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-11-02 12:01:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer):
- Fix potential leaks.
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: fix leak when parsing undefined EXIF tag into tag list
+ gst_buffer_set_data() does not set GST_BUFFER_MALLOCDATA, but
+ the code assumes the buffer takes ownership of the memory
+ allocated earlier.
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
- Fix leak when there is no function configured.
+2010-11-02 11:57:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: fix GstDateTime leak when converting exif tag to tag list
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
- (gst_v4lsrc_buffer_finalize):
- Correctly chain up the finalize method.
+2010-11-01 17:00:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.30.4 pre-release
- * ext/gio/gstgiostreamsink.c:
- * ext/gio/gstgiostreamsrc.c:
- Add documentation and example code for giostreamsink/giostreamsrc.
+2010-11-01 16:59:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/pipelines/gio.c: (GST_START_TEST):
- Ask the GMemoryOutputStream for the data instead of assuming that
- the pointer to the data stayed the same. It could've been realloc'ed.
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/fr.po:
+ * po/ja.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/sl.po:
+ po: update translations
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-30 16:07:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/gio/gstgiosink.c:
- * ext/gio/gstgiosrc.c:
- Make the documentation of giosink/giosrc complete, large parts
- are based on the gnomevfssink/gnomevfssrc docs.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: update docs for discoverer API changes as well
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-30 16:03:18 +0100 Matthias Clasen <mclasen@redhat.com>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- * docs/plugins/gst-plugins-base-plugins.signals:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gdp.xml:
- * docs/plugins/inspect/plugin-gio.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playback.xml:
- * docs/plugins/inspect/plugin-queue2.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-uridecodebin.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- Add the GIO documentation again and while at that run make update.
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ examples: update some more code for new Gtk+ API, with fallback for older Gtk+ versions
+ Move code to new Gtk+ 3.x / 2.9x API. We have defines in place
+ already that make this code work fine on older Gtk+ 2.x.
+ https://bugzilla.gnome.org/show_bug.cgi?id=632653
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-28 15:13:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
- * ext/alsa/gstalsasink.c: (set_swparams):
- * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
- Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
- against libasound >= 1.0.16, since it's been deprecated in
- 0.10.16, and alignment is always 1 then, apparently. (#512899)
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ seek: Define the new combobox API to the old functions if using older GTK
+ https://bugzilla.gnome.org/show_bug.cgi?id=632653
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-30 15:31:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gen_audio_element):
- * gst/playback/gstplaysink.c: (gen_audio_chain):
- Handle case where we can't create the volume element a bit
- better (#514307).
+ * win32/common/libgstutils.def:
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstutils.dsp:
+ win32: remove unused libgstutils stuff
+ Cruft from before the lib was renamed to pbutils
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-28 18:51:08 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
- * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
- Add support for https protocol. Fixes #510229.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * tools/gst-discoverer.c:
+ * win32/common/libgstpbutils.def:
+ discoverer: rename boolean getters for consistency
+ Rename _get_is_image() to _is_image() and _get_interlaced() to _is_interlaced().
+ https://bugzilla.gnome.org/show_bug.cgi?id=633311
-2008-02-11 Julien Moutte <julien@fluendo.com>
-
- Patch by: Alan Peevers <peeves@pacbell.net>
+2010-10-30 12:24:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
- lock when calling alsa methods.
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ pbutils: remove padding from now-private GstDiscovererInfo structure
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-30 12:03:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c:
- Bump rank of jpeg and png typefinders, which will return maximum
- probability in the most common cases (thus short-circuiting more
- expensive typefinders like the mp3 one for these two quite common
- image types).
+ * Makefile.am:
+ * tools/.gitignore:
+ * tools/Makefile.am:
+ tools: rename gst-discoverer binary to gst-discoverer-0.10
+ We're not providing a wrapper like we do for the tools in core,
+ since wrappers are confusing (e.g. for debugging purposes),
+ mostly pointless (since the API is likely to change between
+ major versions), and cause packaging issues when packages for
+ two different major versions are to be installed in parallel.
+ https://bugzilla.gnome.org/show_bug.cgi?id=633023
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-30 11:41:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoraparse.c:
- Fix long description of the theora parser to be more verbose than just
- the type name.
+ * tools/gst-discoverer.c:
+ tools: update gst-discoverer tool for last-minute API change
+ https://bugzilla.gnome.org/show_bug.cgi?id=633311
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-29 14:17:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Branko Čibej <brane at xbc dot nu>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: calculate better timeout value
+ We want to send the keealive message a little earlier than the timeout value
+ specifies. Scale this based on the value of the timeout instead of just assuming
+ 5 seconds.
- * sys/xvimage/xvimagesink.c:
- Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
- Fixes bug #515654.
+2010-10-29 14:24:54 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-2008-02-09 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: don't let the rtsp connection timeout
+ Because we should act before the rtsp server does a timeout, we
+ reduce the timeout-time with 5 seconds, this should be safe to always
+ keep te rtsp connection alive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=633455
- * gst/playback/gstplaybasebin.c:
- Set is_dynamic as True if there are elements with both request
- and sometimes src pad templates instead of breaking out when it
- finds the first pad template that is a src.
+2010-10-28 15:55:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/Makefile.am:
+ * tests/check/elements/videoscale.c:
+ videoscale: Add unit test for working reverse negotiation
+ See bug #633147.
- * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
- (update_streams), (video_combo_cb), (audio_combo_cb),
- (text_combo_cb), (volume_spinbutton_changed_cb), (main):
- Add some stream switching and volume gui for playbin2.
+2010-10-29 11:48:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix wrong flowreturn handling
+ Oggdemux will currently try to pad alloc a buffer from the peer when it is
+ reading the header files. This is a relic from the time where we had an internal
+ parser and needs to be removed at some point in time.
+ The problem is that when there is no peer pad yet (which is normal when
+ collecting headers) we should still continue to parse all the packets of a
+ page instead of erroring out on NOT_LINKED.
+ Fixes #632167
- * gst/playback/gstplay-marshal.list:
- Added marshal for streamselector Tags.
-
- * gst/playback/gstplaybasebin.c: (set_active_source):
- Streamselector now selects pads based on the pad object instead of its
- name.
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (init_group), (gst_play_bin_init), (get_group), (get_tags),
- (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
- (gst_play_bin_get_text_tags),
- (gst_play_bin_set_current_video_stream),
- (gst_play_bin_set_current_audio_stream),
- (gst_play_bin_set_current_text_stream),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
- Remove option to mute streams with the current-a/v/t property, we have
- this functionality in the flags.
- Add signals to notify when the number of A/V/T channels changed.
- Add action signals to get tags for the A/V/T streams.
- Implement setting the current A/V/T stream.
- Rearrange some things to simplify stream selection.
- Implement volume.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
- (gst_play_sink_get_volume), (gst_play_sink_set_property),
- (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
- (activate_vis), (gst_play_sink_reconfigure):
- * gst/playback/gstplaysink.h:
- Add and implement volume setting methods.
-
- * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
- (gst_selector_pad_finalize), (gst_selector_pad_get_property),
- (gst_selector_pad_event), (gst_stream_selector_class_init),
- (gst_stream_selector_init), (gst_stream_selector_finalize),
- (gst_stream_selector_set_property),
- (gst_stream_selector_get_property),
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_request_new_pad):
- * gst/playback/gststreamselector.h:
- Add pad properties for tags and status of pads.
- Keep tags on pads.
- Make active pad selection based on pad object instead of name.
-
-2008-02-08 Stefan Kost <ensonic@users.sf.net>
+2010-10-29 11:47:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Revert last change as we now check in gtk-doc.m4 for sed.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: add some more debug statements
-2008-02-08 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-10-26 16:41:28 +0100 Jan Schmidt <thaytan@noraisin.net>
- * configure.ac:
- Find and subst SED when building the docs.
+ * gst/playback/gstplaysink.c:
+ playsink: Fix subpicture overlay when deinterlacing disabled.
+ Fix a bug when reconfiguring the playsink where the subpicture
+ stream is broken by attempting to connect it through
+ streamsynchroniser and second time.
-2008-02-08 Julien Moutte <julien@fluendo.com>
+2010-10-28 17:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
- (main): Make sure bus signals are reconnected when pressing STOP
- and then PLAY again for a parse launch pipeline. Fix a ref leak
- on the bus.
- * win32/common/config.h: Updated.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ discoverer: use const in most of the getters
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-28 03:09:10 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Make DISABLE_DEPRECATED defined *only* during CVS, not during
- pre-releases or releases.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ discoverer: use specific types in getters
+ Use GstDiscoverer{Audio,Video}Info in getters like
+ gst_discoverer_{audio,video}_info_get_*(). This avoids the casts in the macros,
+ help language bindings and is more correct.
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-28 11:56:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * ext/gio/Makefile.am:
- Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
- reporting
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ discoverer: Move documentation to the correct section
+ And don't mention the (not existing) libgstdiscovery.
+ https://bugzilla.gnome.org/show_bug.cgi?id=633336
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-27 13:16:37 +0100 Jan Schmidt <thaytan@noraisin.net>
- * docs/plugins/Makefile.am:
- Add the headers which need scanning for the GIO plugin. The rest of
- the docs still need migrating.
+ * common:
+ Automatic update of common submodule
+ From 7bbd708 to 011bcc8
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-24 16:09:26 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/Makefile.am:
- * tests/check/Makefile.am:
- * tests/check/pipelines/.cvsignore:
- Add gio in a few more places.
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Get pad caps if we can't get negotiated caps
+ Better provide something than nothing
+ https://bugzilla.gnome.org/show_bug.cgi?id=632988
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-24 15:38:30 +0200 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- * ext/Makefile.am:
- * tests/check/Makefile.am:
- Move gio plugin from -bad and mark as experimental.
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ discoverer: Don't ref a NULL caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=632988
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-09-24 16:02:42 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/interfaces/mixeroptions.c:
- * gst-libs/gst/interfaces/mixertrack.c:
- Comment out a couple of other things which break the build when
- GST_DISABLE_DEPRECATED isn't on but -Werror is.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't add non prerolled stream to topology
+ If a final stream didn't preroll, don't add it to the topology since
+ it doesn't give any information at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=632988
-2008-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-24 16:17:09 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix pbutils header.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Description for RealAudio container format
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-24 15:38:42 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Fix compiler warning.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for VP6 alpha and ASS subtitle
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-22 17:44:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.30.3 pre-release
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
- Clear the addrinfo struct using memset. Fixes #514937.
+2010-10-20 11:01:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ seek: The new combo box text API is available since 2.23.0 and 2.91.1
+ Only use it conditionally.
- * gst/tcp/gstfdset.h:
- Remove unused field to same some memory.
+2010-10-20 11:01:14 +0200 Matthias Clasen <mclasen@redhat.com>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Mark action signals as such.
+ * tests/examples/seek/jsseek.c:
+ seek: Don't use deprecated combo box text API
+ Fixes bug #632653.
-2008-02-06 Michael Smith <msmith@fluendo.com>
+2010-10-21 12:24:19 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/theora/theoradec.c: (_theora_granule_frame),
- (_inc_granulepos):
- Increment granulepos for new-bitstream versions appropriately.
- Fixes #514623.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: workaround internal decodebin2 failing state change
+ Fixes #632656.
-2008-02-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-21 13:38:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek),
- (rate_spinbutton_changed_cb), (update_streams), (main):
- Remove obsolete stream_time reset after flushing seek, core does that
- automatically now.
- Improve accuracy of speed spinbutton.
- Only do playbin2 stuff when we actually use it.
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/icles/test-colorkey.c:
+ * tests/icles/test-xoverlay.c:
+ tests: don't use deprecated gtk_widget_hide_all()
+ gtk_widget_hide_all() has been deprecated in gtk+ 2.x and
+ removed in 2.9x master. Just use gtk_widget_hide() instead.
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-21 13:07:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Revert previous change of the test environment's GST_PLUGIN_PATH.
- The problem is not with the plugins, but with element factories
- and only occurs if elements are split out from existing plugins
- or if plugins change name (see #512740).
+ * tools/Makefile.am:
+ tools: fix linking problems caused by accidentally linking against installed pbutils/gstvideo libs
+ Fixes build errors in jhbuild:
+ /foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_element_factory_list_get_elements'
+ ../gst-libs/gst/pbutils/.libs/libgstpbutils-0.10.so: undefined reference to `gst_element_link_pads_full'
+ /foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_element_factory_list_filter'
+ ../gst-libs/gst/pbutils/.libs/libgstpbutils-0.10.so: undefined reference to `gst_pad_link_full'
+ /foo/build/gst-plugins-base/gst-libs/gst/video/.libs/libgstvideo-0.10.so: undefined reference to `gst_plugin_feature_list_debug'
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-19 00:07:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Fix the tests environment's GST_PLUGIN_PATH: we want the directory
- with the core's plugins first and our local build directories last,
- since we might be building against an installed core, and that
- core's plugin directory may contain older or other versions of
- our own -base plugins, but we really do want to test our local
- ones (if there are multiple plugins or element factories with the
- same name, those inspected last will trump those read in earlier).
- Fixes #512740 for the most part.
+ * gst-libs/gst/pbutils/pbutils-marshal.list:
+ * gst-libs/gst/rtsp/gstrtsp-marshal.list:
+ libs: touch marshal.list files to force rebuild after Makefile.am changes
+ Force regeneration of marshal.[ch] files after prefix changes in
+ Makefile.am, to avoid build errors for those of us who don't
+ habitually make clean first.
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-16 01:08:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
- Use gmtime_r if available as gmtime is not MT-safe.
- Fixes bug #511810.
-
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/pbutils-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.30.2 pre-release
- * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
- Cast glong to time_t as time_t might have a different type on
- other platforms, like FreeBSD, and we get a compiler warning
- otherwise. Fixes bug #511825.
+2010-10-16 01:07:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (get_group), (get_n_pads), (gst_play_bin_get_property),
- (pad_added_cb), (no_more_pads_cb), (perform_eos),
- (autoplug_select_cb), (deactivate_group):
- Remove stream-info, we going for something easier.
- Refactor getting the current group.
- Implement getting the number of audio/video/text streams.
+2010-10-08 17:24:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init), (gst_stream_selector_init),
- (gst_stream_selector_get_property),
- (gst_stream_selector_request_new_pad),
- (gst_stream_selector_release_pad):
- * gst/playback/gststreamselector.h:
- Add property for number of pads.
-
- * tests/examples/seek/seek.c: (set_scale), (update_flag),
- (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
- (text_toggle_cb), (update_streams), (msg_async_done),
- (msg_state_changed), (main):
- Block slider callback when updating the slider position.
- Add gui elements for controlling playbin2.
- Add callback for async_done that updates position/duration.
-
-2008-02-01 Stefan Kost <ensonic@users.sf.net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ * tests/check/libs/tag.c:
+ tag: Adds GST_TAG_CAPTURING_SOURCE
+ Adds a tag to indicate the source/device used for the capture.
+ Already maps it in exif and adds tests.
+ API: GST_TAG_CAPTURING_SOURCE
+ https://bugzilla.gnome.org/show_bug.cgi?id=631773
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- First round of plugin docs cleansups.
+2010-10-08 15:51:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-cdparanoia.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gdp.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml:
- Regenerate.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ * tests/check/libs/tag.c:
+ tag: Adds GST_TAG_CAPTURING_METERING_MODE
+ Adds a tag to inform what mode was used by a camera to calculate
+ the picture capturing exposure
+ Also adds mapping to exif and tests
+ API: GST_TAG_CAPTURING_METERING_MODE
+ https://bugzilla.gnome.org/show_bug.cgi?id=631773
- * ext/ogg/Makefile.am:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- Add header for oggmux. the c-file needs a doc blob still.
+2010-10-08 15:14:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ * tests/check/libs/tag.c:
+ tag: Adds GST_TAG_CAPTURING_SHARPNESS
+ Adds new tag for tagging sharpness processing used
+ when capturing an image. Also maps it in the exif
+ tags.
+ Tests included.
+ API: GST_TAG_CAPTURING_SHARPNESS
+ https://bugzilla.gnome.org/show_bug.cgi?id=631773
- Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+2010-10-15 23:54:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_set_extension_data):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add gst_rtp_buffer_set_extension_data()
- Add a unit test for this addition. Fixes #511478.
- API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ * win32/common/libgstrtsp.def:
+ rtsp: don't export marshaller function
+ Make sure the marshaller function isn't exported. As it was
+ never in a public header file, this should be fine.
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-15 21:22:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Back to CVS
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: fix distcheck
+ Apparently noinst implies dist.
-=== release 0.10.17 ===
+2010-10-15 11:23:02 -0700 David Schleef <ds@schleef.org>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * tests/check/Makefile.am:
+ tests: Don't dist generated orc code
- * configure.ac:
- releasing 0.10.17, "Peanut Butter and Jelly"
+2010-10-15 11:22:45 -0700 David Schleef <ds@schleef.org>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ Update generated orc code
- * gst-libs/gst/interfaces/mixeroptions.c:
- * gst-libs/gst/interfaces/mixertrack.c:
- Also remove the conditional registration of the signals
- that disappeared with the ABI change in 0.10.14
+2010-10-15 19:18:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/pbutils.h:
+ * win32/common/libgstpbutils.def:
+ pbutils: make marshaller private
+ There's no reason to make the marshaller public API. Don't install
+ pbutils-marshal.h header file and use prefix that makes sure the
+ symbol doesn't get exported.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Revert patch to gstrtspconnection.c for brown paper bag
- release of -base. Re-opens: #511825
+2010-10-15 19:14:49 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: use fewer variables in Makefile.am to make things clearer
+ Also fix typo in DISTCLEANFILES.
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/mixertrack.h:
- Change the way these deprecated function pointers are removed
- so that the compiled ABI is unconditionally smaller. This
- sets in stone an ABI break that actually occurred when the
- things were deprecated in 0.10.14, which seems to be the best
- fix as the only known users are oss-mixer and sunaudio-mixer in
- gst-plugins-good.
- Fixes: #513018
+2010-10-15 17:59:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-30 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure: bump Orc requirement to 0.4.11
+ Has fixes for volume, among other things.
- * win32/common/libgstpbutils.def:
- Export the two new _get_type() functions which are needed
- by the python bindings.
+2010-10-15 17:23:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ docs: improve gst_discoverer_new() docs a bit
- * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
- Cast glong to time_t as time_t might have a different type on
- other platforms, like FreeBSD, and we get a compiler warning
- otherwise. Fixes bug #511825.
+2010-10-15 16:43:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: private structs need to padding
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init):
- Initialize the GstRingerBuffer class to get it's debug category
- initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
- category and otherwise we get some g_critical(). Fixes bug #512334.
+2010-10-15 11:26:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/video/video.c:
+ video: Fix stupid copy&paste error in last commit
- * configure.ac:
- Back to CVS
+2010-10-13 22:51:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.16 ===
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add support for A420
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-13 20:45:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- releasing 0.10.16, "Scheduled Interruption"
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: API: Add A420 video format
+ This is planar 4:2:0 YUV plus non-subsampled alpha plane.
-2008-01-22 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-14 12:31:39 -0700 David Schleef <ds@schleef.org>
- Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+ * common:
+ Automatic update of common submodule
+ From 5a668bf to 7bbd708
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_get_extension_data):
- Fix typos and wrong extension check. Fixes #511274.
+2010-10-14 16:36:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-18 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix ADTS caps stream-format detail
+ Field should be "stream-format", not "stream-type".
- * po/sk.po:
- Oops - add new sk.po mentioned in the LINGUAS I just committed
+2010-07-08 15:22:08 +0200 Andrzej K. Haczewski <ahaczewski@gmail.com>
-2008-01-17 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: extend AAC typefinder to detect LOAS streams
+ Extend AAC typefinder to recognize LOAS stream as specified by
+ ISO/IEC 14496-3:2009.
+ https://bugzilla.gnome.org/show_bug.cgi?id=623918
- * po/LINGUAS:
- Add ca translation to the disted list.
+2010-10-13 23:26:35 +0300 Stefan Kost <ensonic@users.sf.net>
- * win32/vs6/libgstsdp.dsp:
- Convert line endings to CRLF
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ *decodebin*: set pad-templates on ghostpads
+ This makes calling gst_pad_get_pad_template() work.
-2008-01-17 Sebastien Moutte <sebastien@moutte.net>
+2010-10-12 21:23:03 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * win32/MANIFEST:
- Add win32/vs6/libgstrtsp.dsp to MANIFEST
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: Update with latest datetime from core
+ Updates datetime functions to latest APIs in core
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-13 16:12:38 +0300 Stefan Kost <ensonic@users.sf.net>
- * win32/common/libgstsdp.def:
- * win32/common/libgstvideo.def:
- Add new API declarations
+ * ext/theora/gsttheoraparse.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstsubparse.c:
+ various: add a missing G_PARAM_STATIC_STRINGS flag to object properties
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-10-13 13:05:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/gsttheoradec.h:
- * ext/theora/gsttheoraparse.h:
- * ext/theora/theoradec.c:
- * ext/theora/theoraparse.c:
- Take a 2nd stab at handling libtheora granulepos changes in the decoder
- and parser by inspecting the bitstream version of the incoming data.
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: declare decodebin2 stable, deprecate the old decodebin
+ https://bugzilla.gnome.org/show_bug.cgi?id=624949
-2008-01-14 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-13 12:55:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-audio-uninstalled.pc.in:
- * pkgconfig/gstreamer-audio.pc.in:
- * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
- * pkgconfig/gstreamer-cdda.pc.in:
- * pkgconfig/gstreamer-fft-uninstalled.pc.in:
- * pkgconfig/gstreamer-fft.pc.in:
- * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
- * pkgconfig/gstreamer-floatcast.pc.in:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-interfaces.pc.in:
- * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
- * pkgconfig/gstreamer-netbuffer.pc.in:
- * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
- * pkgconfig/gstreamer-pbutils.pc.in:
- * pkgconfig/gstreamer-riff-uninstalled.pc.in:
- * pkgconfig/gstreamer-riff.pc.in:
- * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
- * pkgconfig/gstreamer-rtp.pc.in:
- * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
- * pkgconfig/gstreamer-rtsp.pc.in:
- * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
- * pkgconfig/gstreamer-sdp.pc.in:
- * pkgconfig/gstreamer-tag-uninstalled.pc.in:
- * pkgconfig/gstreamer-tag.pc.in:
- * pkgconfig/gstreamer-video-uninstalled.pc.in:
- * pkgconfig/gstreamer-video.pc.in:
- Provide one pkg-config file for every gst-plugins-base library.
- This makes linking to those libraries much more intuitive and
- provides standard pkg-config behaviour for them. Fixes bug #499697.
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ playbin2: declare stable, deprecate the old playbin
+ https://bugzilla.gnome.org/show_bug.cgi?id=624949
-2008-01-13 David Schleef <ds@schleef.org>
+2010-10-12 16:03:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/vs_4tap.c:
- Fix valgrind error on 4tap scaling method.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: only keep last valid granulepos
+ Only keep the last valid granulepos we see when scanning the last
+ pages. It is possible that the last page that we inspect has a -1 granulepos, in
+ which case we want to keep the previous valid time instead.
+ Fixes #631703
-2008-01-13 Sebastien Moutte <sebastien@moutte.net>
+2010-10-10 15:22:52 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
- Include Winsock2.h for VS6 and use a different way initialize
- hints structure so it can build with VS6.
- * win32/MANIFEST:
- * win32/vs6/libgstsdp.dsp:
- * win32/common/libgstsdp.def:
- Add new files for libgstsdp.
- * win32/vs6/grammar.dsp:
- Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstdecodebin.dsp:
- * win32/vs6/libgstdecodebin2.dsp:
- * win32/vs6/libgstplaybin.dsp:
- * win32/vs6/libgstvolume.dsp:
- Add new dependencies to the link list.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix check for last page
-2008-01-13 Julien Moutte <julien@fluendo.com>
+2010-10-10 15:22:04 -0700 David Schleef <ds@schleef.org>
- * win32/common/config.h:
- * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
- (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
- (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
- (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
- (gst_rtsp_header_field_get_type), (gst_rtsp_status_code_get_type):
- * win32/common/interfaces-enumtypes.c:
- (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
- (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
- (gst_mixer_track_flags_get_type),
- (gst_tuner_channel_flags_get_type):
- * win32/common/multichannel-enumtypes.c:
- (gst_audio_channel_position_get_type):
- * win32/common/pbutils-enumtypes.c:
- (gst_install_plugins_return_get_type):
- * win32/common/pbutils-enumtypes.h: Update/Add generated files
- in the win32 build directory.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: change checks from is_skeleton to is_sparse
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-10-10 15:17:31 -0700 David Schleef <ds@schleef.org>
- * tests/check/Makefile.am:
- Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: move is_sparse into stream map
- * tests/check/elements/audiorate.c: (do_perfect_stream_test):
- * tests/check/elements/playbin.c:
- * tests/check/libs/mixer.c: (test_element_interface_supported),
- (gst_implements_interface_init):
- * tests/check/libs/rtp.c: (GST_START_TEST):
- Fix various assignment type mismatches.
+2010-10-11 18:06:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * tests/check/Makefile.am:
+ tests: vorbis: adds missing lib
+ Adds missing lib to vorbis check tests makefile
- * configure.ac:
- * gst-libs/gst/rtsp/Makefile.am:
- Add test to see if hstrerror is available or if we need libresolv
- (Solaris) for it, then use it in libgstrtsp.
+2010-10-11 14:30:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Set GST_ELEMENT_IS_SOURCE flag
+ uridecodebin behaves like a source, let's mark it as a source
- * gst-libs/gst/tag/Makefile.am:
- Fix include path order
+2010-10-10 00:52:13 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-11 Edward Hervey <edward.hervey@collabora.co.uk>
+ * ext/theora/gsttheoradec.c:
+ theoradec: expose telemetry properties only if libtheora was compiled with --enable-telemetry
+ Since this is just a debugging feature and libtheora will usually not be
+ compiled with that option enabled, we should maybe just hide these properties,
+ since they won't work anyway, and avoid confusing warnings.
+ Also rename properties to make them less cryptic.
+ https://bugzilla.gnome.org/show_bug.cgi?id=628488
- * gst-libs/gst/pbutils/install-plugins.c:
- (gst_install_plugins_context_copy),
- (gst_install_plugins_context_get_type):
- * gst-libs/gst/pbutils/install-plugins.h:
- Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
- for bindings.
+2010-10-09 23:49:35 +0100 Alexey Fisher <bug-track@fisher-privat.net>
-2008-01-11 Michael Smith <msmith@fluendo.com>
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoradec.h:
+ theoradec: add properties to enable debugging telemetry overlay
+ The theora decoder can overlay debugging information on the output
+ video. This functionality is only available if libtheora has been
+ compiled with --enable-telemetry. For more details see:
+ http://people.xiph.org/~xiphmont/demo/theora/demo2.html
+ Based on original patch by Michael Smith <msmith at xiph org>
+ https://bugzilla.gnome.org/show_bug.cgi?id=628488
- * ext/theora/theoradec.c: (gst_theora_dec_class_init),
- (_theora_granule_frame), (_theora_granule_start_time),
- (theora_dec_sink_convert), (theora_dec_decode_buffer):
- Adapt for post-alpha meaning of granulepos, when we
- have a newer version of libtheora.
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
- (theora_enc_is_discontinuous), (theora_enc_chain):
- Likewise.
- * tests/check/Makefile.am:
- Link libtheora into theoraenc test so we can check which version of
- libtheora we're testing against.
- * tests/check/pipelines/theoraenc.c: (check_libtheora),
- (check_buffer_granulepos),
- (check_buffer_granulepos_from_starttime), (GST_START_TEST),
- (theoraenc_suite):
- Adapt tests to check the values that are now defined for theora; make
- the tests backwards-adapt the passed values if we're running against an
- old libtheora.
- Fixes #497964
+2010-10-10 18:35:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-10 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Make sure that the caps for upstream negotiation are simple caps
+ Fixes bug #631774.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init):
- Ref audio clock class from a thread-safe context to make sure
- we're not bit by GObjects lack of thread-safety here (#349410),
- however unlikely that may be in practice.
+2010-10-09 14:17:57 +0100 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2008-01-10 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ * tests/examples/app/appsrc-stream.c:
+ * tests/examples/app/appsrc-stream2.c:
+ examples: g_mapped_file_unref exists already since GLib 2.21.3
- * autogen.sh:
- Add -Wno-portability to the automake parameters to stop warnings
- about GNU make extensions being used. We require GNU make in almost
- every Makefile anyway.
-
- * configure.ac:
- Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
- at the same time is required for per target flags.
+2010-10-07 19:32:56 +0200 Guillaume Emont <gemont@igalia.com>
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix seeking with negative rate with skeleton
+ Files with a skeleton, or other files with a stream that ends before the end of
+ the chain would start playing from the end of the chain when trying to seek with
+ a negative rate at a position between the end of any stream and the end of the
+ chain.
+ This is due to the loop in _do_seek() assuming that pages will be encountered
+ for all streams shortly after the place where we want to seek, as found by
+ do_binary_search().
+ In the first iteration of the loop, stream ends are now checked against the
+ time of the current page.
+
+2010-10-07 18:53:35 +0100 Zaheer Abbas Merali <zaheerabbas@merali.org>
- * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
- Post an error message if we can't pull as many bytes as we need
- for the tag. This makes sure the user gets to see a proper error
- message if a file with a partial ID3 tag is fed to decodebin, and
- not a 'no ID3 tag demuxer' error, which would be confusing
- (see #508138).
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: gdp protocol is deprecated. People should use gdppay instead.
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-08 12:43:28 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description strings for ID3, APE, and ICY tags.
+ * common:
+ Automatic update of common submodule
+ From c4a8adc to 5a668bf
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-28 12:17:41 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Make sure we error out correctly if we can't activate one of
- the elements we've added. Fixes #508138.
-
-2008-01-07 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst-libs/gst/pbutils/gstdiscoverer-private.h:
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ pbutils: rename gstdiscoverer-private.h to pbutils-private.h
- Patch by: Bastien Nocera <hadess at hadess net>
+2010-09-28 12:15:22 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
- (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
- Use snd_mixer_selem_set_{playback|capture}_volume_all() if
- the volume is the same for all channels. This works around
- some problem in alsa that leaves us with inconsistent state
- for some reason (#486840).
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst-libs/gst/pbutils/gstdiscoverer-private.h:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ pbutils: Use copy_and_clean_caps for description methods
+ This allows the various _get_*_description() methods to be more
+ forgiving with the provided caps.
-2008-01-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-08 12:51:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Jerone Young <jerone at gmail com>
+ * common:
+ Automatic update of common submodule
+ From 5e3c9bf to c4a8adc
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
- If there's no mixer track by the name of 'Master' or 'Front',
- check if there's one called 'PCM' before trying the generic
- fallback logic (fixes #506928, where we pick 'Mic' as master
- track for the AD1984 card in a Thinkpad T61/X61 laptop).
+2010-10-08 11:23:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-07 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: make public _get_type() functions thread-safe
- * gst/playback/gstplay-enum.c:
- (register_gst_autoplug_select_result),
- (gst_autoplug_select_result_get_type), (register_gst_play_flags),
- (gst_play_flags_get_type):
- * gst/playback/gstplay-enum.h:
- Add enums for configuration flags.
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (no_more_pads_cb),
- (autoplug_select_cb), (gst_play_bin_change_state):
- Merge mode with flags.
- Add more property getters/setters, defaults and docs.
- Add properties to get number of audio/video/text streams.
- Create sink object in _init so that we can always rely on it being
- there.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_init),
- (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
- (activate_vis), (gst_play_sink_reconfigure),
- (gst_play_sink_set_flags), (gst_play_sink_get_flags),
- (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Use flags to configure the sink pipelines.
- Add tee before audio pipeline so that we can use it for visualisations.
- Start working on integrating visualisations.
- Remove mode, we can do everything with the flags now.
- Add method to configue the sink pipeline.
+2010-10-08 10:29:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtspurl: minor clean-up
+ Merge and const-ify two arrays that should be one.
- * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
- (check_buffer_timestamp), (check_buffer_duration):
- Turn these functions into macros so we can see right away
- where the failure occured.
+2010-10-08 10:06:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-05 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsp: fix enum value name in enums that are public API
+ https://bugzilla.gnome.org/show_bug.cgi?id=629746
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
- debugging information to understand how X calculates the stride
- for XvImages.
+2010-10-08 09:48:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audio: make public get_type() functions thread-safe
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (volume_choose_func),
- (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
- (volume_setup):
- * gst/volume/gstvolume.h:
- Use GstAudioFilter as base class for the volume element instead of
- plain GstBaseTransform.
+2010-10-08 09:45:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audio: fix enum value name in enums that are public API
+ So run-time bindings can introspect the names correctly (we abuse this
+ field as description field only in elements, not for public API
+ (where the description belongs into the gtk-doc chunk).
+ https://bugzilla.gnome.org/show_bug.cgi?id=629746
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
- Don't set element details for the abstract GstAudioFilter class.
+2010-10-08 12:30:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-02 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/volume/gstvolumeorc-dist.c:
+ volume: Regenerate generated orc C code again with an orc fix for loading double parameters
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
- Implement get_unit_size() vmethod of GstBaseTransform.
+2010-10-08 11:50:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-01 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ volume: Update generated orc sources
- * gst-libs/gst/pbutils/Makefile.am:
- * gst-libs/gst/pbutils/pbutils.h:
- Use glib-enum generator to have a proper enum GType for
- GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
+2010-10-08 11:49:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Fix unit test failure for the controlled processing functions
+ Going over integer arithmetic will lead to minimal rounding errors,
+ leading to +/-1 changes for volume==1.0. Implement the controlled
+ processing with floating point arithmetic, which was already done
+ for the C versions anyway.
- * tests/check/Makefile.am:
- * tests/check/pipelines/theoraenc.c:
- Reenable theoraenc test, which fails on the buildbot but
- not locally.
+2010-10-08 09:10:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 David Schleef <ds@schleef.org>
+ * configure.ac:
+ configure: Require orc 0.4.10
- * docs/libs/.cvsignore:
- * docs/plugins/.cvsignore:
- Add *-undeclared.txt to fix buildbot.
+2010-10-07 23:54:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ audioconvert: Update generated orc files
- * tests/check/Makefile.am:
- Second attempt at disabling theoraenc test long enough to
- get buildbot to compile -base.
+2010-10-07 23:54:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst/volume/gstvolumeorc.orc:
+ volume: Update for orc changes
+ double parameters are declared with .doubleparam now.
- * tests/check/pipelines/theoraenc.c:
- Disable theoraenc test long enough to get the buildbot to
- compile a recent -base.
+2010-10-03 11:21:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ volume: Update generated orc sources
- * tests/examples/seek/seek.c: (stop_cb):
- Make sure we reset the slider value to 0.0 without racing against a
- possible g_idle that sets it to something else.
+2010-10-03 12:00:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/volume/gstvolumeorc.orc:
+ volume: Fix controlled processing via orc
- * sys/ximage/ximagesink.c:
- fix typo
+2010-10-03 11:24:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-30 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/volume/gstvolume.c:
+ volume: Actually enable usage of the orc optimized functions
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- Add Location header so that we can start implementing redirects.
- See #506025.
+2010-10-03 11:20:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Implement int32 processing with orc
- * gst/subparse/gstssaparse.c:
- combine if's
+2010-10-01 12:21:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Implement controlled processing for int16/1-2ch and int8/1,2,4ch with orc
- * gst/subparse/gstssaparse.c:
- remove duplicate log message
-
-2007-12-29 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/libvisual/visual.c: (gst_visual_chain):
- Fix 'xyz may be used uninitialized' compiler warnings caused
- by broken g_assert_not_reached() macro in GLib-2.15.x and don't
- abort() in any case but properly report the error.
-
-2007-12-28 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (gst_play_bin_finalize), (gst_play_bin_set_uri),
- (gst_play_bin_set_suburi), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
- (autoplug_select_cb), (activate_group), (deactivate_group),
- (setup_next_source), (save_current_group),
- (gst_play_bin_change_state):
- Code cleanups.
- Remove next-uri, we can use the uri property just fine.
- Fix some crasher.
- Unref uridecodebin when switching.
- Fix going to READY.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
- (gst_play_sink_init), (gst_play_sink_dispose),
- (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
- (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
- (gst_play_sink_set_property), (gst_play_sink_get_property),
- (gen_video_chain), (gen_text_element), (gen_audio_chain),
- (gen_vis_element), (gst_play_sink_get_mode),
- (gst_play_sink_set_mode), (gst_play_sink_set_flags),
- (gst_play_sink_get_flags), (gst_play_sink_request_pad),
- (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
- (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Add some locking to make things threadsafe.
+2010-10-01 11:13:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/test7.c: (about_to_finish_cb):
- Fix test.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Implement controlled processing for f64/1ch and f32/1-2ch in orc
-2007-12-22 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-01 11:00:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
- (gst_video_scale_get_property), (gst_video_scale_transform_caps),
- (gst_video_scale_transform):
- Don't claim to be able to handle/transform caps that can't really
- be handled by the currently selected scaling method (here: RGB or
- packed YUV with 4-tap method). Also add locking to method property.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Convert parts of the controlled processing to orc
- * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
- (test_basetransform_based):
- Some test pipelines for the above (not entirely valgrind clean yet
- apparently).
+2010-10-01 10:44:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-21 David Schleef <ds@schleef.org>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Implement f64 scaling with orc
+ This requires orc 0.4.10
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Add additional RGBA and RGB-24 video formats.
+2010-10-01 10:38:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ audioconvert: Implement remaining conversion functions from/to doubles to orc
+ This requires orc 0.4.10
- * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
- (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
- (test_suburi_error_wrongproto), (test_missing_primary_decoder):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
- (cddabasesrc_suite):
- Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
- deprecated in the future (see #498924).
+2010-10-07 20:54:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: use g_object_notify_by_pspec() if possible
+ Use g_object_notify_by_pspec() when building against GLib >= 2.26.
+ This avoids the pspec lookup which takes the global paramspec pool lock.
- * gst/playback/gststreamselector.c: (gst_selector_pad_event):
- Don't leak event.
+2010-10-07 20:37:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-20 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: use g_object_notify_by_pspec() if possible
+ Use g_object_notify_by_pspec() when building against GLib >= 2.26.
+ This avoids the pspec lookup which takes the global paramspec pool lock.
- * gst-libs/gst/riff/riff-read.c:
- Use GST_ROUND_UP_2 macro
+2010-10-04 10:01:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Do not advance segment starts beyond stop times
+ Advance stop times too when they are getting higher than the
+ stop time of segments, avoiding assertions.
+ The stop time has to be advanced too so that running time keep in sync
+ for gapless mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=631312
- * gst/playback/.cvsignore:
- Ignore more.
+2010-10-06 16:19:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/libs/rtp.c:
+ tests: rtp: No need to unref buffer from bufferlist
+ Buffers obtained from buffer list iterators don't need to
+ be unreffed.
+ Test was failing due to this.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
- * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
- (set_active_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (setup_sinks), (playbin_set_subtitles_visible):
- Make switching off of subtitles work. To avoid all kind of
- problems with unlinking of the subtitle input, we just keep
- the subtitle inputs linked as they are and tell textoverlay
- not to render them. Fixes #373011.
- Other subtitle switching issues (esp. when there are both
- external and in-stream subtitles) remain. They'll be solved
- in playbin2.
+2010-10-04 11:22:45 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/vorbis/gstvorbisdec.c:
+ vorbisdec: reverse playback; decode pending buffers upon EOS
- * gst/playback/gststreamselector.c: (gst_selector_pad_init):
- Init the pad segment too.
+2010-10-05 19:15:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: use math-compat.h here as well
+ Hopefully the powers that be don't mind the gst/glib include here
+ too much.
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
- (gst_audioringbuffer_open_device),
- (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release), (gst_audioringbuffer_start),
- (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
- (gst_audio_sink_create_ringbuffer):
- Improve debug output.
+2010-10-05 19:13:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (gst_ring_buffer_delay):
- Prevent some functions from doing things and failing when the
- ringbuffer is not yet acquired.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: include new math-compat.h header for rint() on MSVC
+ Should fix compilation with Visual Studio 2008.
+ https://bugzilla.gnome.org/show_bug.cgi?id=630802
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-05 17:19:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/interfaces.h:
- Also remove interfaces.h from CVS as it is not needed anymore.
+ * win32/common/libgstrtp.def:
+ win32: update def file with new RTP methods
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+2010-10-05 17:13:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/Makefile.am:
- interfaces.h is not used anymore so remove it from the build
- process.
+ * tests/check/libs/rtp.c:
+ check: fix rtp checks
+ Fix the checks for the extension support in RTP.
-2007-12-17 David Schleef <ds@schleef.org>
+2010-10-05 16:36:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- Add a "blink" pattern. Turn on the pain. Apologies. It's useful
- for testing vertical refresh synchronization.
+ * tests/examples/seek/seek.c:
+ seek: fix position reporting
-2007-12-17 David Schleef <ds@schleef.org>
+2010-08-26 12:34:11 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Add new GstVideFormat enum and write a bunch of helper functions
- based around it.
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ rtcpbuffer: Add function to manipulation the data in RTCP feedback packets
+ Add methods to get/set the length of the Feedback Control Information (FCI) as
+ well as getting a pointer to the FCI itself.
-2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-23 16:41:44 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * Makefile.am:
- Use new common/win32.mak.
+ * tests/check/libs/rtp.c:
+ tests: Test the manipulations of bufferlists containing RFC 5285 header extensions
-2007-12-17 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-08-23 14:24:21 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Add debug info.
- When going from PLAYING to PAUSED, pause the ringbuffer before calling
- the parent state change function, just like the audiosink, because the
- parent waits for the element to finish its processing before completing
- the state change. This makes going to PAUSED a lot snappier.
- When going from READY to PAUSED, don't allow the ringbuffer to start
- yet.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add function to transform a GstBuffer into a GstBufferList
+ Add a new function called gst_rtp_buffer_list_from_buffer() that takes
+ a GstBuffer containing a RTP packets and spits out a GstBufferList
+ containing two buffers, one with the header and the other with the payload.
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-08-22 19:44:38 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Yet another fix for broken software that produce files with an empty
- blockalign field. Instead of completely failing, make a second attempt
- at guessing the width/depth by looking at strf->size.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add functions to add RFC 5285 header extensions to GstBufferLists
+ Add functions to add header extensions to buffer lists, these functions only modify
+ the header part of the buffer lists, so the data is not copied.
-2007-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-22 17:22:21 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
- (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
- * gst-libs/gst/pbutils/install-plugins.c:
- (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_plugin_message_get_installer_detail),
- (gst_missing_encoder_installer_detail_new):
- * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Turn a few g_assert_not_reached() into g_return_val_if_reached() to
- avoid compiler warnings (#503930).
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add function to read RFC 5285 header extensions from GstBufferLists
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-08-20 15:30:08 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
- for jpeg video streams.
- Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
- for the above modification.
+ * tests/check/libs/rtp.c:
+ tests: Add test for RTP header extension functions
-2007-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-20 17:13:06 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
- (gst_x_overlay_handle_events):
- More guards (we don't want klass to end up being NULL).
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add function to add RTP header extensions with a two bytes header
-2007-12-14 Sebastian Dröge <slomo@circular-chaos.org>
+2010-08-20 12:54:38 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * configure.ac:
- * gst/volume/gstvolume.c: (gst_volume_init):
- Use new gst_base_transform_set_gap_aware() function as volume
- correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
- for this.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add function to append RFC 5285 one byte header extensions
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-08-19 16:26:18 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * tests/examples/seek/seek.c: (msg_segment_done), (main):
- Don't go to READY on EOS as this avoids testing of seeking and
- restarting after EOS, use the stop button when you want to READY.
- Don't try to do a flushing seek in segment-done, it does not make
- sense to use this for gapless playback and is not needed.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ rtpbuffer: Add function to parse RFC 5285 header extensions
+ RFC 5285 describes a generic method to add multiple header extensions to RTP packets.
+ These functions parse these headers and return them, both for the one-byte header and the
+ two bytes headers.
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-05 12:05:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
- (reset_rate_timer), (update_in_rates), (update_out_rates),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_chain), (gst_queue_loop):
- Use separate timers for input and output rates.
- Pause measuring the output rate when we block for more data.
- See #503262.
+ * ext/libvisual/visual.c:
+ libvisual: only drop frames that are really too old
+ Also take the frame duration into account so that we don't drop frames that are
+ partially past the estimated QoS time.
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-05 12:01:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_chain):
- Pause the timer to measure the input rate when we block because the
- queue is filled. See #503262.
+ * ext/libvisual/visual.c:
+ libvisual: add latency query
+ Add our own latency to the latency query reply from upstream.
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-10-05 12:00:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * ext/libvisual/visual.c:
+ libvisual: add some defines
+ Add some defines for width/height/fps and a define for the minimum amount of
+ samples we need to buffer.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_free):
- Close control sockets. Fixes #503440.
+2010-10-04 15:48:51 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tools/gst-discoverer.c:
+ gst-discoverer: The 'ready' signal was renamed to 'finished'
- * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
- Expose the right pad in the right place with the right element.
+2010-10-04 17:27:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoraparse.c:
+ parse: Don't error on discont
+ We don't need to error out when we detect a discontinuity.
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description for 'private' dts caps (who come up with that name?).
+2010-10-04 17:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoraparse.c:
+ theoraparse: set caps on streamheader too
- * Makefile.am:
- Add check-exports target and run it with 'make check'.
+2010-10-04 13:07:14 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * configure.ac:
- Be stricter about what we export in our libraries: change regexp so that
- we only export _gst_foo(), but not __gst_foo().
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ build: Fix include path order for gir generation
+ This makes sure that the built girs are picked up over installed girs
+ where this is currently the case.
- * gst-libs/gst/cdda/base64.h: (rfc822_binary):
- * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
- Change internal functions to __gst_foo so they dont' get exported.
+2010-10-01 14:52:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add missing symbols.
+ * gst-libs/gst/pbutils/codec-utils.c:
+ codec utils: populate mpeg4 caps "level" field with level, not profile
+ Call the right function to get the level. Also add some more debug
+ logging.
-2007-12-11 David Schleef <ds@schleef.org>
+2010-10-01 10:47:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ChangeLog: remove conflict markers
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ volume: Update generated orc files
-2007-12-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-10-01 10:42:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
- Use gst_tag_freeform_string_to_utf8() here, which also takes
- into account any character sets specified by the user via
- environment variables.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ volume: Improve f32 scaling by using only a single array
+ Passing the same array as dest and src is invalid anyway because
+ they're maked with the restrict qualifier.
-2007-12-10 Wim Taymans <wim.taymans@gmail.com>
+2010-09-30 15:19:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/Makefile.am:
- Also link to libm.
+ * gst-libs/gst/pbutils/codec-utils.c:
+ pbutils: include config.h in codec utils
-2007-12-10 Wim Taymans <wim.taymans@gmail.com>
+2010-09-30 00:19:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- No need for floating point operations here. avoids having to link
- against the math library too.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/codec-utils.c:
+ docs: add new codec utils API to docs
-2007-12-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-05-01 01:03:18 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (formats),
- (format_info_get_desc):
- * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
- (GST_START_TEST):
- Add one or two missing formats. Generate ADPCM description
- dynamically depending on layout/format.
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ * win32/common/libgstpbutils.def:
+ pbutils: Add MPEG-4 Video profile/level extraction
+ This adds code to translate the profile_and_level indication from the
+ MPEG-4 video (ISO/IEC 14496-2) headers to a string profile/level. The
+ mappings are taken from the spec and Wireshark's code, and might need to
+ be expanded on.
+ https://bugzilla.gnome.org/show_bug.cgi?id=617314
+ API: gst_codec_utils_mpeg4video_get_profile()
+ API: gst_codec_utils_mpeg4video_get_level()
+ API: gst_codec_utils_mpeg4video_caps_set_level_and_profile()
+
+2010-04-30 20:50:09 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ pbutils: add H.264 profile/level extraction functions to codec utils
+ This adds code to parse the first few bytes of H.264 sequence parameter
+ set in order to extract the profile and level as const strings. This
+ code was originally in both qtdemux and matroskademux.
+ https://bugzilla.gnome.org/show_bug.cgi?id=617314
+ API: gst_codec_utils_h264_get_level()
+ API: gst_codec_utils_h264_get_profile()
+ API: gst_codec_utils_h264_caps_set_level_and_profile()
+
+2010-04-30 15:12:04 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ * win32/common/libgstpbutils.def:
+ pbutils: add AAC profile detection to codec utils
+ This moves AAC profile detection to pbutils, and uses this in
+ typefindfunctions. This will also be used in qtdemux.
+ https://bugzilla.gnome.org/show_bug.cgi?id=617314
+ API: gst_codec_utils_aac_get_profile()
+ API: codec_utils_aac_caps_set_level_and_profile()
-2007-12-09 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-30 13:41:17 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * configure.ac:
- Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/codec-utils.c:
+ * gst-libs/gst/pbutils/codec-utils.h:
+ * gst-libs/gst/pbutils/pbutils.h:
+ * gst/typefind/Makefile.am:
+ * gst/typefind/gstaacutil.c:
+ * gst/typefind/gstaacutil.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ * win32/common/libgstpbutils.def:
+ pbutils: add codec-specific utility functions for AAC
+ This allows us to add generic codec-specific functionality, like
+ extracting profile/level data from headers, without having to duplicate
+ code across demuxers and typefindfunctions.
+ As a starting point, this moves over AAC level extraction code from
+ typefindfunctions, so it can be reused in qtdemux, etc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=617314
+ API: gst_codec_utils_aac_get_sample_rate_from_index()
+ API: gst_codec_utils_aac_get_level()
+
+2010-09-30 13:12:30 +0300 René Stadler <rene.stadler@nokia.com>
-2007-12-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/tags.c:
+ tags: fix unused function warning with debug disabled
- Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
+2010-09-30 12:59:46 +0300 René Stadler <rene.stadler@nokia.com>
- * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
- Some .srt files start with chunk number 0 and not chunk number 1,
- recognise and accept those as well (fixes #502497).
+ * gst-libs/gst/tag/tags.c:
+ tags: fix illegal use of internal debug category function
+ From gstinfo.h:
+ /* do not use this function, use the GST_DEBUG_CATEGORY_INIT macro */
+ GstDebugCategory *_gst_debug_category_new (const gchar * name,
+ And more importantly:
+ #pragma GCC poison _gst_debug_category_new
+ So this commit fixes --disable-gst-debug builds.
- * tests/check/elements/subparse.c: (srt_input), (srt_input0),
- (test_src):
- Add unit test for the above.
+2010-09-29 18:57:50 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-12-06 Wim Taymans <wim.taymans@gmail.com>
+ * tools/gst-discoverer.c:
+ gst-discoverer: Print out topology if available.
+ If we have some unhandled streams, we can still print out the remaining
+ topology.
- * gst/playback/gstplay-enum.c:
- (register_gst_autoplug_select_result),
- (gst_autoplug_select_result_get_type):
- * gst/playback/gstplay-enum.h:
- Add missing files.
+2010-09-29 18:54:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-12-05 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't post async-done when not needed
+ Where it was previously located, we would get async-done for the first
+ unknown-type, even if other valid streams would appear afterwards.
+ decode_bin_expose() will take care of posting async-done when the group
+ is exposed.
+ But we still want to post it in case the typefinding returned an unknown
+ type, in which case we will post it after posting an error.
+ These two changes ensure we do as much as possible before posting async-done.
- * gst/playback/Makefile.am:
- Group decodebin2 and uridecodebin into the same plugin so that they
- can share the GEnumType.
-
- * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
- (_gst_select_accumulator), (gst_decode_bin_class_init),
- (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
- (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
- (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
- Add signal to sort factories instead of the more awkward autoplug-select
- signal.
- Modify autoplug_select so that we can try, skip or expose the
- autopluggin of an element on a pad.
-
- * gst/playback/gstfactorylists.c: (compare_ranks),
- (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
- (element_filter), (gst_factory_list_get_elements),
- (gst_factory_list_debug), (gst_factory_list_filter):
- * gst/playback/gstfactorylists.h:
- Simplify the API, allow getting elements based on mask.
+2010-09-29 16:53:21 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst/playback/gstplay-marshal.list:
- Add some more marshallers.
-
- * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
- (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
- (autoplug_select_cb), (activate_group):
- Add support for managing non-raw sinks by providing a custom element and
- sink list to decodebin2.
- Try to plug non-raw sinks when decodebin2 using autoplug-select of
- decodebin2.
-
- * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
- (gst_play_sink_set_mode), (gst_play_sink_request_pad):
- * gst/playback/gstplaysink.h:
- Add support for raw and non-raw sinks.
- Add support to force sinks selected by playbin2.
- Don't plug raw converters for non-raw sinks.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepay: ensure metadata is writable
- * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
- (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
- (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
- (plugin_init):
- Use right accumulators.
- Proxy new signal.
+2010-09-29 13:29:20 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add descriptions for more codecs
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- Use runnning time as the base time instead of the timestamp.
- Spotted by Saur on IRC.
+2010-09-29 12:33:44 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-12-03 Edward Hervey <bilboed@bilboed.com>
+ * tests/examples/seek/seek.c:
+ seek: Fix debug statement argument type
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
+2010-09-28 09:30:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+ * tests/check/Makefile.am:
+ * tools/Makefile.am:
+ More makefile Fixes
+ Removing some not needed lines added in the last makefile
+ fixes commit (previous commit).
+ Also adds some more makefile files to check tests
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_read_chain):
- If we find a new serial number but it does not contain a BOS page, make
- sure we initialize the chain to NULL because else we will try to scan it
- and crash. Fixes #500763
+2010-06-17 14:32:22 +0300 René Stadler <rene.stadler@nokia.com>
-2007-11-30 Wim Taymans <wim.taymans@gmail.com>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: allow render rectangle coordinates to be negative
+ Useful for cropped zooming.
- * gst/playback/Makefile.am:
- * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
- (get_feature_array), (decoders_filter), (sinks_filter),
- (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
- (gst_factory_list_filter):
- * gst/playback/gstfactorylists.h:
- Refactor some common code to filter factories and check caps compat.
+2010-06-17 14:33:44 +0300 René Stadler <rene.stadler@nokia.com>
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (gst_decode_bin_autoplug_continue),
- (gst_decode_bin_autoplug_factories),
- (gst_decode_bin_autoplug_select), (analyze_new_pad),
- (find_compatibles):
- * gst/playback/gstplaybin.c:
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (gst_play_bin_finalize),
- (autoplug_factories_cb), (activate_group):
- * gst/playback/gstqueue2.c:
- * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
- (proxy_autoplug_continue_signal),
- (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
- (proxy_drained_signal):
- Add some more debug info and use factor filtering code.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xoverlay: allow render rectangle coordinates to be negative
+ This is useful for cropped zooming of the overlay.
-2007-11-26 Stefan Kost <ensonic@users.sf.net>
+2010-09-28 15:15:57 +0300 René Stadler <rene.stadler@nokia.com>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/volume/gstvolume.c:
- * gst/volume/gstvolume.h:
- Add GAP-flag support.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xoverlay: fix endless loop in deprecated method
-2007-11-24 Julien MOUTTE <julien@moutte.net>
+2010-09-28 08:46:25 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): Increase the range of the
- rate selector as I would like to test QOS behavior at higher
- forward and reverse playback speed like say 64x.
+ * tests/examples/app/Makefile.am:
+ * tools/Makefile.am:
+ Fixing Makefiles
+ Adds some missing lines to makefiles
-2007-11-23 Sebastian Dröge <slomo@circular-chaos.org>
+2010-09-27 18:14:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (audioresample_query), (audioresample_query_type),
- (gst_audioresample_set_property):
- Implement latency query.
+ * gst-libs/gst/tag/tags.c:
+ tags: add debug category for tags utility functions
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2010-09-27 14:36:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain):
- Our EOS time contains the base_time, _wait_eos() expects a running_time
- so we have to subtract the base_time again before calling the function.
- This fixes an EOS regression where the base_time was added twice and EOS
- took longer and longer in certain situations.
- Fixes #498767.
+ * gst-libs/gst/tag/tags.c:
+ tags: try ISO-8859-1 as second fallback in case WINDOWS-1252 is not supported
+ Better safe than sorry. Some embedded systems may use crippled iconv
+ implementations or not support WINDOWS-1252 for other reasons.
+ https://bugzilla.gnome.org/show_bug.cgi?id=630471
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2010-09-23 23:53:48 +0300 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
- (gst_base_audio_sink_set_provide_clock),
- (gst_base_audio_sink_get_provide_clock),
- (gst_base_audio_sink_set_slave_method),
- (gst_base_audio_sink_get_slave_method),
- (gst_base_audio_sink_set_property),
- (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
- (gst_base_audio_sink_none_slaving),
- (gst_base_audio_sink_handle_slaving):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Expose methods for some object properties so that subclasses can more
- easily configure them.
- Added slave method none, that completely disables slaving to the
- internal clock.
- API: gst_base_audio_sink_set_provide_clock()
- API: gst_base_audio_sink_get_provide_clock()
- API: gst_base_audio_sink_set_slave_method()
- API: gst_base_audio_sink_get_slave_method()
+ * gst-libs/gst/tag/tags.c:
+ tags: when converting freeform strings try Windows-1252 as fallback instead of ISO-8859-1
+ Windows-1252 is a superset of ISO-8859-1, which uses some space
+ allocated to control characters for additional printable characters.
+ https://bugzilla.gnome.org/show_bug.cgi?id=630471
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_set_provide_clock),
- (gst_base_audio_src_get_provide_clock),
- (gst_base_audio_src_set_property),
- (gst_base_audio_src_get_property), (gst_base_audio_src_create):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- Expose methods for some object properties so that subclasses can more
- easily configure them.
- API: gst_base_audio_src_set_provide_clock()
- API: gst_base_audio_src_get_provide_clock()
+2010-09-24 21:30:20 -0700 David Schleef <ds@schleef.org>
-2007-11-20 Wim Taymans <wim.taymans@gmail.com>
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: ptalarbvorm speed level goes to 3
- Patch by: Joe Peterson <lavajoe at gentoo dot org>
+2010-09-24 16:31:37 +0200 Vladimir <wl2776 at gmail.com>
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix compilation on FreeBSD (Gentoo). Fixes #498228.
+ * tests/examples/seek/seek.c:
+ seek: Add #define for seekbar granularity
+ Fixes #630496
-2007-11-19 Sebastian Dröge <slomo@circular-chaos.org>
+2010-09-24 14:03:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/libs/fft.c: (GST_START_TEST):
- Fix scaling to really have dB instead of something else.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/audio-enumtypes.c:
+ * win32/common/audio-enumtypes.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/gstrtsp-enumtypes.h:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/pbutils-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ win32: define GST_PACKAGE_RELEASE_DATETIME in win32 config.h as well
+ Also update enums.
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2010-09-24 00:25:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): There's a nice macro to check
- GTK version, use it.
+ * tests/check/elements/.gitignore:
+ .gitignore: ignore new appsrc unit test
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2010-09-24 13:09:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): Try to support stable version
- of GTK.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: add Since markers
+ Fixes #630443
-2007-11-17 Stefan Kost <ensonic@users.sf.net>
+2010-07-30 13:54:42 +0200 Havard Graff <havard.graff@tandberg.com>
- * gst/playback/README:
- * gst/playback/test7.c:
- Fix the build + little README update.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * win32/common/libgstaudio.def:
+ baseaudiosink: Added getter and setter for drift tolerance.
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2010-09-24 12:54:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
- Add playbin2 seek pipeline.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: subtract the render_delay from our latency
+ The latency reported by the base class includes the render_delay, which we don't
+ want to include when we start slaving our clocks.
+ See #630441
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2010-09-23 23:57:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstplayback.c: (plugin_init):
- * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
- (eos_cb), (about_to_finish_cb), (main):
- Add playbin2.
- Added gapless playback example.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: Use G_DEFINE_ABSTRACT_TYPE instead of manual GObject boilerplate code
+ This also makes the _get_type() function threadsafe.
+ Fixes bug #630440.
- * gst/playback/gstplaybasebin.c:
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
- * gst/playback/gstqueue2.c:
- * gst/playback/test.c:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
- (pad_removed_cb):
- * gst/playback/gststreaminfo.h:
- Change email.
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_uri),
- (gst_play_bin_set_suburi), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gst_play_bin_handle_message),
- (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
- (drained_cb), (unlink_group), (activate_group),
- (setup_next_source), (gst_play_bin_change_state),
- (gst_play_bin2_plugin_init):
- Added raw first version of playbin2. Does chained oggs and gapless
- playback fine. No support for raw sinks yet. No visualisations or
- subtitles yet.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
- (gst_play_sink_class_init), (gst_play_sink_init),
- (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
- (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
- (gst_play_sink_set_property), (gst_play_sink_get_property),
- (post_missing_element_message), (free_chain), (add_chain),
- (activate_chain), (gen_video_chain), (gen_text_element),
- (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
- (gst_play_sink_set_mode), (gst_play_sink_request_pad),
- (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
- (gst_play_sink_send_event), (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Added Element that abstracts the sinks and their pipelines for playbin2.
-
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
- (gst_selector_pad_class_init), (gst_selector_pad_init),
- (gst_selector_pad_finalize), (gst_selector_pad_reset),
- (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
- (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
- (gst_selector_pad_chain), (gst_stream_selector_get_type),
- (gst_stream_selector_base_init), (gst_stream_selector_class_init),
- (gst_stream_selector_init), (gst_stream_selector_set_property),
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_getcaps),
- (gst_stream_selector_is_active_sinkpad),
- (gst_stream_selector_activate_sinkpad),
- (gst_stream_selector_get_linked_pads),
- (gst_stream_selector_request_new_pad),
- (gst_stream_selector_release_pad):
- * gst/playback/gststreamselector.h:
- Improve streamselector, make it select and unselect the current pad more
- intelligently.
- Subclass GstPad for the sinkpads of the selector.
- Handle segments more correctly.
- Fix caps negotiation.
- Implement release_pad.
-
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
- (gst_decode_group_check_if_drained), (source_pad_event_probe),
- (remove_fakesink):
- Add drained signal fired when decodebin finishes decoding the data.
- Remove deprecated STATE_DIRTY message.
-
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
- (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
- (analyse_source), (proxy_drained_signal), (make_decoder),
- (source_new_pad), (value_list_append_structure_list),
- (handle_redirect_message), (handle_message):
- Proxy the new drained signal.
- Handle pad removed from decodebin.
- Handle redirect messages by sorting multiple redirections based on the
- connection speed.
-
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
-
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
-
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Fix leaking headers. Fixes #496761.
-
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
-
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_change_state):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
- Don't leak the PAR on errors. Fixes #496731.
-
-2007-11-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
- (gst_tag_from_id3_user_tag):
- Add mapping for audio cd discid tags, so we can extract
- them from tags as well (see #347848). Also compare identifiers
- in ID3v2 TXXX frames in a case-insensitive way to increase
- compatibility when reading tags (discid vs. DiscID vs. DiscId).
+2010-09-23 10:16:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tags: exif: Add mapping for _HORIZONTAL_ERROR
+ Maps GST_TAG_GEO_LOCATION_HORIZONTAL_ERROR to the
+ GPSHPositionError tag in exif.
+ Tests included.
- * gst-plugins-base.doap:
- Oops, fix the release name.
+2010-09-22 14:10:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/app/gstappsrc.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/appsrc.c:
+ appsrc: Do not override buffer caps if appsrc caps is null
+ Make appsrc not set caps on buffers when its own caps is NULL.
+ This avoids calling make_metadata_writable on all buffers and
+ prevents losing buffer caps in case we are not replacing it
+ with something meaningful.
+ https://bugzilla.gnome.org/show_bug.cgi?id=630353
- * gst-plugins-base.doap:
- Add 0.10.15 release
+2010-09-21 18:57:42 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraenc.h:
+ theoraenc: Make the bitrate/quality dynamically modifiable
+ https://bugzilla.gnome.org/show_bug.cgi?id=630303
- * configure.ac:
- Back to CVS
+2010-09-22 12:35:59 +0200 Edward Hervey <bilboed@bilboed.com>
-=== release 0.10.15 ===
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * tools/gst-discoverer.c:
+ discoverer: Fixup DiscovererResult handling
+ This was a leftover from the changes from a flag to an enum
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2010-09-22 12:10:24 +0200 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- releasing 0.10.15, "No need to argue"
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: We don't need the signals from the queues
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2010-09-22 01:50:21 -0700 David Schleef <ds@schleef.org>
- * win32/vs6/libgstfft.dsp:
- Convert line endings to DOS.
+ * gst-libs/gst/Makefile.am:
+ gst-libs: build pbutils after video
+ Because pbutils now depends on video.
-2007-11-15 Sebastien Moutte <sebastien@moutte.net>
+2010-09-21 18:33:36 +0200 Edward Hervey <bilboed@bilboed.com>
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstfft.dsp:
- * win32/MANIFEST:
- Add a project file for fft plugin and remove socket
- based plugin which don't build from the workspace.
- * win32/vs6/libgstaudio.dsp:
- * win32/vs6/libgstrtp.dsp:
- * win32/vs6/libgsttag.dsp:
- Convert line endings back to DOS.
- Fixes #496724
+ * common:
+ Automatic update of common submodule
+ From aa0d1d0 to 5e3c9bf
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-09-20 21:04:48 +0300 Stefan Kost <ensonic@users.sf.net>
- * win32/vs6/libgstinterfaces.dsp:
- * win32/vs6/libgstrtsp.dsp:
- Convert line endings back to DOS
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ discoverer: fix docs
+ While the doc parser allows for certain variation, it is a good idea to not
+ use random characters here and there, but try to stick to the little markup
+ syntax there is.
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-09-20 16:45:32 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/fft/kiss_fft_f32.h:
- * gst-libs/gst/fft/kiss_fft_f64.h:
- * gst-libs/gst/fft/kiss_fft_s16.h:
- * gst-libs/gst/fft/kiss_fft_s32.h:
- Don't include malloc.h which doesn't exist on Mac OSX.
- Instead, pull in glib.h and use g_malloc/g_free for
- consistency. Fixes: #496548
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Fix debug statement.
+ Fixes build on macosx
-2007-11-09 Stefan Kost <ensonic@users.sf.net>
+2010-09-20 16:28:52 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstdecodebin2.c:
- Dont leak ghostpad. Fixes #475451.
+ * gst/volume/gstvolumeorc-dist.c:
+ volume: orc fixup for loading float arguments
+ This is only used with DISABLE_ORC.
-2007-11-09 Wim Taymans <wim.taymans@gmail.com>
+2010-09-20 11:24:10 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/design/design-decodebin.txt:
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- Update some more docs and comments.
+ * tools/.gitignore:
+ * tools/Makefile.am:
+ * tools/gst-discoverer.c:
+ tools: Standalone tool for discovering media file properties
+ Fixes #625944
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-20 11:23:36 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: Sebastien Moutte <sebastien moutte net>
+ * win32/common/libgstpbutils.def:
+ win32: Update with symbols from GstDiscoverer
+ Fixes #625944
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
- (gst_rtcp_unix_to_ntp):
- * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
- Fix some C99-isms and and a missing function that some versions of
- MSVC don't like too much (#494346).
+2010-09-20 11:23:17 +0200 Edward Hervey <bilboed@bilboed.com>
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstaudio.dsp:
- * win32/vs6/libgstrtp.dsp:
- * win32/vs6/libgsttag.dsp:
- Update vs6 projects files (#494346).
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ docs: Documentation for new pbutils GstDiscoverer
+ Fixes #625944
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-20 11:22:32 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/pbutils/.gitignore:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/gstdiscoverer-private.h:
+ * gst-libs/gst/pbutils/gstdiscoverer-types.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ * gst-libs/gst/pbutils/gstdiscoverer.h:
+ * gst-libs/gst/pbutils/pbutils-marshal.list:
+ * gst-libs/gst/pbutils/pbutils.h:
+ pbutils: New Discoverer utility
+ Fixes #625944
- * win32/common/libgstaudio.def:
- * win32/common/libgstcdda.def:
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstnetbuffer.def:
- * win32/common/libgstpbutils.def:
- * win32/common/libgstrtp.def:
- * win32/common/libgstrtsp.def:
- * win32/common/libgsttag.def:
- * win32/common/libgstvideo.def:
- More missing symbols to export (fixes #493986).
+2010-09-20 11:13:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-11-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add mp3 to the apetag extensions
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/fft/gstfftf32.c:
- * gst-libs/gst/fft/gstfftf32.h:
- * gst-libs/gst/fft/gstfftf64.c:
- * gst-libs/gst/fft/gstfftf64.h:
- * gst-libs/gst/fft/gstffts16.c:
- * gst-libs/gst/fft/gstffts16.h:
- * gst-libs/gst/fft/gstffts32.c:
- * gst-libs/gst/fft/gstffts32.h:
- * tests/check/libs/fft.c: (GST_START_TEST):
- Remove the magnitude and phase calculation functions as these have
- very special use cases and can't even be used for the spectrum
- element. Also adjust the docs to mention some properties of the used
- FFT implemention, i.e. how the values are scaled. Fixes #492098.
+2010-09-18 13:15:08 -0700 David Schleef <ds@schleef.org>
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix regression in ball pattern
+ Was painting using two different methods.
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
- (finish_source):
- Avoid crash when there are external subtitles (fixes #491722).
+2010-09-17 11:46:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-11-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Fix ACCEPTABLE_CHAR() macro to work with signed and unsigned chars
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
- 'Could not open resource for writing' is not an acceptable
- error message when we can't open the audio device (see #492334),
- even less so when we're trying to open it to record something.
+2010-09-17 11:44:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Revert "sdp: Remove useless check in macro"
+ This reverts commit e6a041b69fd21c42651d98cf8a3064e43cecc51c.
+ It's not a useless check, the signedness of "char" and "gchar" is
+ defined by the ABI.
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+2010-09-17 10:43:04 +0200 Edward Hervey <bilboed@bilboed.com>
- * win32/common/libgstrtp.def:
- Add some more missing symbols (#492813).
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Remove useless check in macro
+ A signed char is always < 128. Fixes a warning on macosx build.
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-16 18:03:23 -0700 David Schleef <ds@schleef.org>
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/adder/gstadderorc-dist.c:
+ * gst/adder/gstadderorc-dist.h:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.h:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ orc: update generated files to fix MSVC compile issues
- * tests/check/elements/audioconvert.c: (verify_convert):
- Add check to make sure that the out caps have a channel layout
- set on them where they should have one.
+2010-09-16 17:56:31 -0700 David Schleef <ds@schleef.org>
-2007-11-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/gstvideoscaleorc.orc:
+ videoscale: Don't use broken orc feature
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+2010-09-16 19:30:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
- * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
- Include our own _stdint.h instead of sys/types.h, makes MingW happy
- (#492306).
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xoverlay: G_GUINTPTR_FORMAT is since 2.22
+ Don't rely on too new symbols, we only depend on 2.20.
- * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
- Use _pipe directly, GLib doesn't have a pipe() macro any longer
- (it disappeared in GLib 2.14.0) (#492306).
+2010-09-16 15:01:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/sdp/Makefile.am:
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix includes and LIBS for win32/Mingw (#492306).
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/playrec/.gitignore:
+ * tests/examples/playrec/Makefile.am:
+ * tests/examples/playrec/playrec.c:
+ examples: add synchronized playback and capture example
+ Add an example that demonstrates synchronized playback and capture.
- * tests/examples/dynamic/addstream.c (pause_play_stream):
- Use more portable g_usleep() instead of sleep() (#492306).
+2010-09-16 17:15:32 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-2007-11-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Fix indentation
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+2010-09-16 17:14:20 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
- (gst_ring_buffer_parse_caps):
- Return NULL instead of an enum that happens to be 0, fixes warning
- on MSVC (#492114).
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add bar pattern
+ Simple bar with foreground color on the background color
- * gst-libs/gst/audio/gstringbuffer.h:
- No trailing commas in enum list (for gcc-2.9x).
+2010-09-16 15:07:15 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst/videotestsrc/videotestsrc.c: (random_char):
- Make information loss explicit instead of implicitly truncating to
- eight bits via the return value. Fixes runtime error on MSVC when
- using the debug CRT (#492114).
+ * tests/check/elements/videotestsrc.c:
+ tests: use gst-check API in videotestsrc
+ use gst_check_drop_buffers in videotestsrc to
+ clear the global buffers list.
- * win32/common/config.h.in:
- Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
+2010-09-16 14:55:55 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Export a few more symbols (#492114).
+ * tests/check/elements/videotestsrc.c:
+ tests: Fix unit test of videotestsrc
+ Use UYVY for unit tests, it's exactly the same as Y422.
+ (which is currently disabled in videotestsrc)
-2007-11-01 Stefan Kost <ensonic@users.sf.net>
+2010-09-15 15:13:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- Readd the deprecation guards, but preserve compilability.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update for new strings
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-15 15:12:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
- (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
- Preserve channel layout when fixating the number of channels in the
- output caps, or make sure there's a suitable channel position layout
- set on the caps if required. Fixes #430677.
+ * gst-libs/gst/video/video.h:
+ docs: add Since: comment to docs for new GST_VIDEO_FORMAT_UYVP
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-14 11:20:42 -0400 Tristan Matthews <le.businessman@gmail.com>
- * tests/check/elements/decodebin.c: (test_text_plain_streams):
- Make sure the pipeline really operates in push mode as it should
- in this case.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfsrc: set GST_PARAM_MUTABLE_READY flag on the "handle" property
+ Fixes #629672
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-15 15:19:04 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst-libs/gst/audio/audio.h:
- Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
- compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
- (ie. normal cvs builds) will fail.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: fix segfault on negative horizontal-speed
-2007-10-31 Stefan Kost <ensonic@users.sf.net>
+2010-09-15 14:15:13 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/Makefile.am:
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- * gst-libs/gst/interfaces/mixer.c:
- tell gtk-doc about the deprecation guard. Apply more doc fixes.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Actually dispose the unused ghostpads
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-15 11:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/audio.c: (init_value_to_channel_layout),
- (test_channel_layout_value_intersect), (audio_suite):
- Add simple unit test to make sure GstValue intersection
- of channel layouts works the way I think it does.
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ Revert "audioresample: Add GAP flag support"
+ This reverts commit 129af0d8e6a74e8edef3e77c3626616b674b7cc1.
+ This shouldn't be committed at all, it isn't ready and apparently
+ was in the wrong branch locally.
-2007-10-30 Stefan Kost <ensonic@users.sf.net>
+2010-09-15 11:26:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix the docs according to what gtk-doc complained about.
+ * gst-libs/gst/video/convertframe.c:
+ * gst-libs/gst/video/video.h:
+ * tests/check/libs/video.c:
+ video: Add a destroy notify parameter to gst_video_convert_frame_async()
+ Binding generators apparently need this as they can't really know
+ that the callback is guaranteed to be called exactly once and that
+ the user_data can be freed at the end of it.
-2007-10-30 Stefan Kost <ensonic@users.sf.net>
+2010-09-14 12:00:39 +0200 Leo Singer <lsinger@caltech.edu>
- * tests/icles/stress-playbin.c:
- Fix the build.
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ audioresample: Add GAP flag support
+ Fixes bug #586570.
-2007-10-30 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-05 15:17:47 -0700 David Schleef <ds@schleef.org>
- * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- Post nice/more useful error message if we don't have a decoder for
- the primary type.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ * sys/v4l/gstv4lxoverlay.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/examples/overlay/qt-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.cpp:
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ * tests/icles/stress-xoverlay.c:
+ * tests/icles/test-colorkey.c:
+ * tests/icles/test-xoverlay.c:
+ * win32/common/libgstinterfaces.def:
+ xoverlay: Add guintptr versions of functions
+ And deprecate the gulong versions. This is to support platforms
+ where sizeof(unsigned long) < sizeof(void *). Fixes #627565.
+ API: Add gst_x_overlay_set_window_handle()
+ API: Deprecate: gst_x_overlay_set_xwindow_id()
+ API: Add gst_x_overlay_got_window_handle()
+ API: Deprecate: gst_x_overlay_got_xwindow_id()
+ API: Add GstXOverlay::set_window_handle()
+ API: Deprecate: GstXOverlay::set_xwindow_id()
+
+2010-09-14 12:31:58 -0700 David Schleef <ds@schleef.org>
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add UYVP
- * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
- Be a bit more useful, unblock the pads after we fired the no-more-pads
- signal so that we can use the signal to inspect and connect all pads
- without having to keep extra state outside of decodebin.
+2010-09-12 20:36:19 -0700 David Schleef <ds@schleef.org>
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Various improvements
+ Replace moving-color-bars pattern with smpte100, and change
+ moving-speed to horizontal-speed. Default is now 0. Add
+ a rotation stage to pattern building.
+ Allocate a temporary scanline for building images. Remove
+ unused code. Disable several patterns that we're unable to
+ test and probably never used. Add other variants of bayer
+ sampling. Convert some patterns to use videotestsrc_blend_line.
- * gst/playback/gsturidecodebin.c:
- (gst_uri_decode_bin_autoplug_continue),
- (gst_uri_decode_bin_class_init), (no_more_pads_full):
- Implement default signal handler so that we return TRUE when nothing is
- connected.
+2010-09-10 18:10:40 -0700 David Schleef <ds@schleef.org>
-2007-10-28 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: clean up blink pattern
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_wavext_add_channel_layout),
- (gst_riff_wave_add_default_channel_layout),
- (gst_riff_wavext_get_default_channel_mask),
- (gst_riff_create_audio_caps):
- Use the ALSA channel layout as default for wav files without channel
- layout information. This fixes playback of chan-id.wav on 5.1 systems
- for example. Also refactor the channel layout setting a bit and add
- more default channel orders. Fixes #489010.
-
-2007-10-26 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
- GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
- -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
- instead.
+2010-09-10 15:57:54 -0700 David Schleef <ds@schleef.org>
-2007-10-25 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
- (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
- (gst_decode_bin_set_subs_encoding),
- (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
- (gst_decode_bin_get_property), (analyze_new_pad):
- Move subtitle encoding property to decodebin2 so that it can set the
- property value on all elements that it autoplugs and that require it.
- Make caps refcounting more consistent in get/set.
-
- * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
- (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
- (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
- (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
- (proxy_autoplug_continue_signal),
- (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
- (make_decoder):
- Proxy properties and relevant signals from the internal decodebin.
- Make properties MT safe.
-
-2007-10-25 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
- * gst-libs/gst/tag/tags.c:
- Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
- GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Clean up the RGB code
- * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
- Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
+2010-09-10 14:40:44 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
- (gst_tag_to_vorbis_comments):
- Map new SORTNAME tags (these tags aren't even semi-official, so I'm
- just mapping everything I found in the wild) (#414539).
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Convert to intermediate AYUV/ARGB
+ Scanlines are generated into AYUV/ARGB, then converted to the
+ various formats.
-2007-10-24 Wim Taymans <wim.taymans@gmail.com>
+2010-09-10 12:48:30 -0700 David Schleef <ds@schleef.org>
- Inspired by patch of: René Stadler <mail at renestadler dot de>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: rearrange code to work on scanlines
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
- (gst_decode_bin_autoplug_continue),
- (gst_decode_bin_autoplug_factories),
- (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
- (find_compatibles):
- * gst/playback/gstplay-marshal.list:
- Remove the autoplug-sort signal and replace it with a binding friendly
- autoplug-select signal.
- Add an autoplug-factories signal that can be used to generate a list of
- factories to try to autoplug.
- Add the GstPad to the autoplugging signal args as it might be needed to
- make a good factory selection.
- Fix up the marshallers for this. Fixes #407282.
+2010-09-10 12:03:07 -0700 David Schleef <ds@schleef.org>
-2007-10-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix recent breakage of smpte75 pattern
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't abort with an assertion if we receive a seek event with
- a start type of NONE (see launchpad bug #155878).
+2010-09-01 15:18:31 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-2007-10-22 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add moving color bars pattern
+ This pattern is moving the color bars with a given
+ speed. Negative speed is inverting the moving direction.
+ https://bugzilla.gnome.org/show_bug.cgi?id=628500
- * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_change_state), (gst_ximagesink_reset):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
- Make sure that before we clean up the X resources, we shutdown and join
- the event thread.
- Also make sure the event thread does not shut down immediatly after
- startup because the running variable is not yet correctly set.
- Fixes #378770.
+2010-06-14 15:42:09 -0700 David Schleef <ds@schleef.org>
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: refactor using more Orc code
+ Convert downsampling to Orc. Convert horizontal linear scaling
+ to Orc. Combine horizontal and vertical scaling into one pass.
- * gst/playback/gstdecodebin.c: (new_pad), (type_found):
- Make the window for a race in typefind and shutting down smaller until
- we figure out the right locking here. Avoids #485753 usually.
+2010-09-12 19:34:28 -0700 David Schleef <ds@schleef.org>
- * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
- Remove unneeded lock causing a race in typefind and shutting down.
- Fixes #485753.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add UYVP, 10-bit 4:2:2
- * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
- Also remove sinks when going to NULL because we might not complete the
- state change to PAUSED, causing the PAUSED->READY state change not to
- happen.
+2010-09-14 08:41:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ video: Add gst_video_convert_frame_async() to the docs
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
- Also explicitly release the ringbuffer when going to NULL because it
- is required in the setcaps function, before the state change to PAUSED
- completes.
+2010-09-14 08:40:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-16 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstvideo.def:
+ win32: Add gst_video_convert_frame() and gst_video_convert_frame_async() to the .def files
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/stress-playbin.c:
- Does what it says on the tin.
+2010-09-14 08:40:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-15 Wim Taymans <wim.taymans@gmail.com>
+ * tests/check/libs/video.c:
+ video: Add unit test for gst_video_convert_frame_async()
- * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
- Fix queue negotiation. See #486758.
+2010-09-14 08:39:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst-libs/gst/video/convertframe.c:
+ * gst-libs/gst/video/video.h:
+ video: Add async variant of the convert frame function
+ API: gst_video_convert_frame_async()
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
- (gst_xvimagesink_xwindow_new),
- (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
+2010-09-12 16:53:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Fix handling of some of the X atoms. If the last parameter is True,
- XInternAtom won't create the atom if it doesn't exist, and therefore
- might return None. This causes X errors on Xv implementations that
- don't provide the colour balance attributes.
+ * tests/check/libs/video.c:
+ video: Add a simple unit test for the new convert_frame API
+ Unfortunately this can't test the encoding because there's no
+ image encoder in base.
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-12 16:51:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c:
- Extract vorbis comment LICENSE tags correctly.
+ * gst-libs/gst/video/convertframe.c:
+ video: Strip framerate from the target caps
+ There will always be only a single output buffer and if the
+ target caps have a different framerate than the input there
+ will be a negotiation error during conversion.
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-09-12 16:36:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Jason Kivlighn <jkivlighn gmail com>
+ * gst-libs/gst/video/convertframe.c:
+ video: Refactor convert_frame a bit and fix some minor memory leaks in error cases
- * gst-libs/gst/tag/gstid3tag.c:
- * tests/check/libs/tag.c:
- Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
+2010-09-09 14:11:52 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-10-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ * gst/playback/gstscreenshot.c:
+ * gst/playback/gstscreenshot.h:
+ playback: Switch to using gst_video_convert_frame
+ https://bugzilla.gnome.org/show_bug.cgi?id=629157
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't error out when a buggy downstream element doesn't
- handle the newsegment event we send properly (especially
- not without posting a meaningful error message on the
- bus). See bug #471370 and launchpad bug #136264.
+2010-09-09 13:44:54 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-10-10 Wim Taymans <wim.taymans@gmail.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/convertframe.c:
+ * gst-libs/gst/video/video.h:
+ video: Add new method for converting a video frame
+ https://bugzilla.gnome.org/show_bug.cgi?id=629157
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain):
- Use new basesink method to make our EOS drain interruptable.
+2010-09-13 10:02:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-10-10 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: prevent another race with shutdown state change
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Fix silly search-replace oversight.
+2010-09-11 14:55:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-09 Wim Taymans <wim.taymans@gmail.com>
+ * win32/common/libgstsdp.def:
+ win32: Add new SDP symbols to the .def files
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+2010-09-10 18:42:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
- (gst_basertppayload_set_outcaps):
- Fix caps memleak. Fixes #484989.
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: remove leftover g_print
+2010-09-10 17:55:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ sdp: add methods to convert between uri and message
+ Add methods to convert between uri and sdpmessages, loosly based on
+ http://tools.ietf.org/html/draft-fujikawa-sdp-url-01
+ API: GstSDPMessage::gst_sdp_message_parse_uri
+ API: GstSDPMessage::gst_sdp_message_as_uri
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
- Fix debug output.
+2010-09-10 10:40:52 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * tests/check/elements/videotestsrc.c:
+ tests: videotestsrc change the pattern property for the tests
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Also handle the case where there is no clock set on the audio source,
- like in the unit tests.
+2010-09-10 08:42:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-08 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/adder/gstadderorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ orc: Fix generated source files
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
- to avoid compiler warnings
+2010-09-09 20:45:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * tests/examples/seek/seek.c:
+ tests: fix passing of URIs containing '*' and '?' to the seek example
+ Only do wildcard expansion (why?!) on things that look like local
+ file paths. Fixes passing of URIs containing '*' and '?' (see #629212).
- * gst/playback/gstdecodebin.c: (type_found),
- (gst_decode_bin_change_state):
- * gst/playback/gstdecodebin2.c: (type_found),
- (gst_decode_bin_change_state):
- Don't disconnect the have_type signal because we never reconnect it
- later on. Instead keep a variable to see if we already detected a type.
+2010-09-09 21:51:18 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * tests/check/Makefile.am:
+ * tests/check/generic/states.c:
+ tests: allow running state tests for all elements
+ Now one can use GST_NO_STATE_IGNORE_ELEMENTS=1 make generic/states.check
+ to try elements that would normaly be skipped.
- * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
- (type_found):
- Unlink the signal handler when we found the type, we're not going to do
- anything sensible with more type_found signals anyway.
+2010-09-09 11:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ adder: Do debug category initialization in plugin_init again
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't leak caps.
+2010-09-09 10:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadderorc-dist.c:
+ * gst/adder/gstadderorc-dist.h:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.h:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ orc: Update generated source files everywhere
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/gsttagdemux.c:
- * gst-libs/gst/tag/gsttagdemux.h:
- API: add GstTagDemux base class for simple tag demuxers.
+2010-09-09 10:57:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Add GstTagDemux to docs.
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadderorc.orc:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ * gst/audioconvert/plugin.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrcorc.orc:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ Revert "Revert "Use init functions for Orc code""
+ This reverts commit 93aa13639d74449dc68296427e5dbcfe8aca5f51.
+ Everything should work now after regenerating the disted source files.
-2007-10-05 Sebastian Dröge <slomo@circular-chaos.org>
+2010-09-07 19:04:23 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_get_payload_subbuffer):
- Fix bug introduced with last commit which inverted the logic and
- caused all buffers to be dropped. Fixes #483620.
- Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
+ * win32/common/libgstaudio.def:
+ win32: Add new symbol to libgstaudio
-2007-10-04 Stefan Kost <ensonic@users.sf.net>
+2010-09-07 18:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Replace g_return_if_val (as it could be disabled), with regular return
- and warning.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudio: avoid taking extra ref on sink/src
+ Don't take an extra ref on the sink and source because that creates a reference
+ cycle. Instead, use the invalidate method of the clock when the sink and source
+ are freed. This way, we don't call into the time function anymore after the
+ objects are disposed.
-2007-10-03 Stefan Kost <ensonic@users.sf.net>
+2010-09-07 18:06:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/pipelines/simple-launch-lines.c:
- Print message name and not just number.
-
-2007-10-02 Wim Taymans <wim.taymans@gmail.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- When slaved to the clock, don't try to align a sample with the previous
- one when going to PLAYING again.
-
-2007-10-02 Stefan Kost <ensonic@users.sf.net>
-
- * tests/examples/snapshot/snapshot.c:
- Fix the build.
-
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
-
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Add RFC 3556 bandwidth modifiers.
-
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
-
- * docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Update documentation.
-
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
-
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
- (gst_rtp_payload_info_for_name):
- * gst-libs/gst/rtp/gstrtppayloads.h:
- Added new file and header to deal with payload info.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ audioclock: add a function to invalidate the clock
+ Add a function to invalidate the time function of a clock. Useful for when the
+ function becomes invalid.
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
- (gst_rtp_buffer_default_clock_rate):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Payload specific stuff is move to new headers.
- Implement _default_clock rate using the new payload function.
+2010-09-07 16:26:56 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
- (gst_sdp_parse_line):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Add some more comments.
+ * tests/check/Makefile.am:
+ check: Fix linking order of libs/tag
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2010-09-07 16:26:30 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
- (sdp_check_header), (sdp_type_find), (plugin_init):
- Add typefind function for application/sdp.
- Remove some old dirac typefind code that was ifdeffed out.
+ * tests/check/gst-plugins-base.supp:
+ check: Make fontconfig leak suppression more generic
-2007-09-29 Sebastien Moutte <sebastien@moutte.net>
+2010-09-07 08:46:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add new exported functions.
- * win32/vs6/grammar.dsp:
- Add autogeneration and copy of some autegenerated files from win32/common
- for rtsp library.
- * win32/vs6/libgstaudioconvert.dsp:
- Add gstaudioquantize.c to the build.
- * win32/vs6/libgstinterfaces.dsp:
- Add videoorientation.c to the build.
- * win32/vs6/libgstriff.dsp:
- Add libgsttag to the link libraries list.
- * win32/vs6/libgstvolume.dsp:
- Add liboil to the link.
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstrtsp.dsp:
- * win32/common/libgstrtsp.def:
- Add files to build libgstrtsp library.
-
-2007-09-26 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds mappings for new image ppi tags
+ Adds mappings for GST_TAG_IMAGE_HORIZONTAL/VERTICAL_PPI into
+ our exif lib
+ Tests included.
+ Fixes #626570
- * tests/examples/snapshot/snapshot.c: (main):
- Print error when pipeline failed to construct.
+2010-09-07 08:22:27 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tags: Add GST_TAG_IMAGE_HORIZONTAL/VERTICAL_PPI tags
+ Adds new tags for representing the intended PPI of images/videos
+ API: GST_TAG_IMAGE_HORIZONTAL_PPI
+ API: GST_TAG_IMAGE_VERTICAL_PPI
+ Fixes #626570
- * configure.ac:
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Add mappings for the new GST_TAG_COMPOSER for vorbis comments
- and ID3v2 tags.
+2010-09-07 11:41:52 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From c2e10bf to aa0d1d0
- * gst-libs/gst/floatcast/floatcast.h:
- Don't include config.h in an installed public header, this
- might break compilation of applications that don't have such
- a header and doesn't necessarily do what it's supposed to do
- anyway (ie. check for the lrint/lrintf defines) (#442065).
- Add docs for the various macros and document how this header
- has to be used (link against libm, etc.); add a few FIXMEs;
- include math.h for non-c99 code path. Based on patch by
- Jan Schmidt.
-
-2007-09-25 Sebastian Dröge <slomo@circular-chaos.org>
+2010-09-06 18:17:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
- of duplicating these macros in configure.ac.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ rtp: improve basertpdepayload's error message when no input caps were set
+ This is pretty much an FAQ, so try to make the error message a bit
+ more helpful. Also, don't tell people to file a bug in bugzilla
+ about this (which is what happens if the default error message for
+ CORE_NEGOTIATION is used).
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-09-06 13:14:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * po/hu.po:
- * po/sv.po:
- * po/uk.po:
- Updated translations to 0.10.14
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppayload: notify of first timestamp/seqnum
+ Notify of the first timestamp/seqnum pushed out by the payloader.
+ Fixes #612264
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-09-06 11:53:35 +0200 Edward Hervey <bilboed@bilboed.com>
- translated by: Jakub Bogusz <qboosh@pld-linux.org>
+ * gst/videotestsrc/.gitignore:
+ videotestsrc: .gitignore new generate_sine_table
- * po/pl.po:
- Added Polish translation.
+2010-09-06 11:44:17 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstinputselector.h:
+ * gst/playback/gstplaybin2.c:
+ playback: ref the selector pad class inside input-selector
+ Minimizes the delta to original element in -bad and allows us to keep the
+ type static.
- translated by: Ilkka Tuohela <hile@iki.fi>
+2010-09-05 20:57:48 -0700 David Schleef <ds@schleef.org>
- * po/fi.po:
- Added Finnish translation.
+ * gst/videotestsrc/Makefile.am:
+ * gst/videotestsrc/generate_sine_table.c:
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Use static sine table
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-09-05 20:35:13 -0700 David Schleef <ds@schleef.org>
- translated by: Jorge González González <aloriel@gmail.com>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Add foreground/background-color properties
+ Replace solid-color property with foreground-color and add
+ background-color. Pull some common code out of each of the
+ pattern generating functions. Fix many of the patterns to
+ use foreground-color/background-color instead of white/black.
+ Generated images are indentical to previously if foreground-color
+ and background-color are left as default.
+ API: GstVideoTestSrc::foreground-color
+ API: GstVideoTestSrc::background-color
- * po/es.po:
- Added Spanish translation.
+2010-09-05 18:58:03 -0700 David Schleef <ds@schleef.org>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ Automatic update of common submodule
+ From d3d9acf to c2e10bf
- translated by: Mogens Jaeger <mogens@jaeger.tf>
+2010-09-05 17:04:31 -0700 David Schleef <ds@schleef.org>
- * po/da.po:
- Added Danish translation.
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: deprecate colorspec property
+ Fixes: #616392.
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-09-05 12:57:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- translated by: Funda Wang <fundawang@linux.net.cn>
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ audioconvert: Simplify float->s32 conversion
+ orc 0.4.7 is doing saturated conversion from floats to integers
+ and it's not necessary to do this manually anymore.
- * po/zh_CN.po:
- Added Chinese (simplified) translation.
+2010-09-05 12:14:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ Automatic update of common submodule
+ From ca1c867 to d3d9acf
- translated by: Alexander Shopov <ash@contact.bg>
+2010-09-05 12:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/bg.po:
- Added Bulgarian translation.
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ audioconvert: Update disted orc files
-2007-09-21 Wim Taymans <wim.taymans@gmail.com>
+2010-09-05 12:09:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_push_one):
- Fix compilation wrt printf arguments.
+ * gst/volume/gstvolume.c:
+ volume: Enable float processing with orc again
-2007-09-20 Wim Taymans <wim.taymans@gmail.com>
+2010-09-05 12:08:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
- * tests/examples/Makefile.am:
- * tests/examples/snapshot/.cvsignore:
- * tests/examples/snapshot/Makefile.am:
- * tests/examples/snapshot/snapshot.c: (main):
- Add simple snapshot example program using appsink.
+ configure: Require orc 0.4.8.1 for the volume test
-2007-09-20 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/generic/states.c:
- Improved state change unit test.
+2010-08-26 19:16:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ audioconvert: Use the ORC double support
- * docs/plugins/.cvsignore:
- * tests/check/.cvsignore:
- Ignore registries in any format.
+2010-09-04 09:06:08 +0200 Leo Singer <lsinger@caltech.edu>
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ exiftag: Fix compiler warnings with old gcc versions
+ Old gcc complains about possibly uninitialized variables which
+ are always initialized before usage in reality. Fixes bug #628747.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_set_gst_timestamp):
- Only copy timestamp on outgoing packets if the depayloader did not set
- one.
- Also copy duration on outgoing packets.
+2010-08-06 11:53:38 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstfactorylists.c:
+ * gst/playback/gstfactorylists.h:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playback: Switch to gstfactorylist from core
+ https://bugzilla.gnome.org/show_bug.cgi?id=626181
- * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
- (gst_basertppayload_set_outcaps):
- Fix compilation because of missing %d in printf.
- When fixating caps, fixate what we can and throw away all remaining
- unfixed caps, subclasses should do something smart if they need to.
+2010-09-02 12:57:42 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: fix typo in property description
- * ext/gnomevfs/gstgnomevfssrc.c:
- Improve debug logs a bit and be more verbose if things go wrong.
+2010-09-01 17:52:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-17 Jan Schmidt <Jan.Schmidt@sun.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Add support for AYUV
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
- (gst_text_overlay_set_property):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
- (gst_rtcp_unix_to_ntp):
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
- * gst/playback/gstqueue2.c:
- * tests/examples/seek/seek.c: (set_scale):
- Fix a bunch of compile warnings shown with Forte.
+2010-09-01 11:37:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/audiorate/gstaudiorate.c:
- Always pull in config.h before including any system headers.
-
-2007-09-17 Wim Taymans <wim.taymans@gmail.com>
+ audiorate: Fill segment until the end on EOS
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_locked_flush), (gst_queue_locked_enqueue),
- (gst_queue_handle_sink_event), (gst_queue_chain),
- (gst_queue_push_one), (gst_queue_sink_activate_push),
- (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
- Also fix #476514 for queue2.
+2010-09-01 11:33:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-16 Wim Taymans <wim.taymans@gmail.com>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_handle_sink_event),
- (gst_base_rtp_depayload_push_full),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_change_state):
- Remove code to deal with RTP to GST time conversion, we now just copy
- the GST timestamp we receive to the outgoing buffers.
- Handle segment and flushes correctly.
-
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- When we have no valid input timestamp, use the previous rtp timestamp on
- the outgoing RTP packet instead of the RTP base time.
-
-2007-09-15 David Schleef <ds@schleef.org>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fill the segment on EOS or at least produce enough frames to use the complete buffer duration
+ Fixes bug #628400.
- * ext/alsa/gstalsa.c:
- * ext/alsa/gstalsadeviceprobe.c:
- * ext/alsa/gstalsamixer.c:
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasrc.c:
- Change alsa alloca's to malloc to fix warnings on gcc-4.2.
+2010-09-01 11:22:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-15 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Don't ignore flow returns when filling the previous segment during NEWSEGMENT handling
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
- Add some debug info when negotiating caps.
+2010-09-01 11:11:47 +0200 Philippe Normand <pnormand@igalia.com>
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+ * tests/examples/seek/seek.c:
+ seek: allow seeking behind the currently downloaded position.
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- A buffer with an empty payload is also a valid buffer.
+2010-09-01 10:06:09 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+ * gst/adder/gstadder.c:
+ adder: use GST_BOILERPALTE macro
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
- (gst_basertppayload_change_state):
- Make sure we start our RTP timestamp from the random base RTP
- timestamp even if the buffer timestamp starts from some random value.
+2010-08-31 10:09:51 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaysink.c:
+ playback: Set queues silent property to TRUE
+ We don't use the queue signals within playsink.
- * configure.ac:
- * tests/examples/Makefile.am:
- * tests/examples/dynamic/.cvsignore:
- * tests/examples/dynamic/Makefile.am:
- * tests/examples/dynamic/addstream.c: (create_stream),
- (pause_play_stream), (message_received), (eos_message_received),
- (perform_step), (main):
- Add simple exmple app to demonstrate starting and pausing live and
- non-live bins in a PLAYING pipeline.
+2010-08-30 14:59:22 -0500 Rob Clark <rob@ti.com>
-2007-09-14 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: fix Cb/Cr inversion for colored text overlays
+ In case of odd values for xpos or ypos, the division by two in CbCr
+ plane would result in an off-by-one error, which in the case of NV12,
+ NV21, or UYVY would cause inversion of blue and red colors. (And
+ would be not so easily noticed for I420 as it would just cause the
+ chroma to be offset slightly from the luma.)
+ This patch also fixes a silly typo from the earlier patch which
+ added NV12 support that broke UYVY support.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
- typefind for QCP files (RFC #3625)
+2010-08-30 15:50:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-13 Wim Taymans <wim.taymans@gmail.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't reset the pad when pushing resulted in NOT_LINKED
+ The pad might be linked later and after resetting it it will
+ only work after resetting all of oggdemux.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_init):
- Disable pull mode scheduling, we're not ready for it yet and it subtly
- breaks a lot of things.
+2010-08-27 20:45:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Only enable progressive downloading if the upstream duration in bytes is known
+ Otherwise we might try to enable it for live streams, where this would
+ cause playback to fail completely.
+ Fixes bug #628028.
- * tests/check/elements/libvisual.c:
- Test all libvisual plugins, not just the first one; this reproduces
- bug #450336 quite easily. Looks like a problem with the 'jess'
- visualisation.
+2010-08-27 17:23:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't use GST_FLOW_IS_FATAL()
+ And while we're at it, handle WRONG_STATE as error too
+ in oggdemux and WRONG_STATE and NOT_LINKED in oggaviparse.
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/libvisual.c:
- Add basic libvisual test case in an attempt to reproduce bug #450336.
- Doesn't reproduce that bug, but some other crasher instead (invalid
- free), at least with make elements/libvisual.forever and the bumscope
- plugin on x86-64/gutsy. Leaving test disabled for now.
+2010-08-27 11:49:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-11 Wim Taymans <wim.taymans@gmail.com>
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadderorc.orc:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ * gst/audioconvert/plugin.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrcorc.orc:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ Revert "Use init functions for Orc code"
+ This reverts commit b2051090b43f82b23bb01826f09053479bbd7874.
+ Fixes the build again until someone pushes the regenerated .c/.h
+ files too.
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2010-08-22 23:01:19 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
- (read_body), (gst_rtsp_connection_receive):
- Make sure we can not cancel in the middle of receiving a message.
- Fixes #475731.
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: clean up code
+ Merge various color structures into one.
-2007-09-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-22 22:16:45 -0700 David Schleef <ds@schleef.org>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Add ball pattern
- * gst/playback/gstplaybasebin.c:
- Increase upper limit for audio queue a bit; fixes preroll problem
- with playbin and decodebin2 when playing a quicktime trailer with
- multichannel audio via http (#464666).
+2010-08-19 15:43:09 -0700 David Schleef <ds@schleef.org>
-2007-09-10 Wim Taymans <wim.taymans@gmail.com>
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadderorc.orc:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ * gst/audioconvert/plugin.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrcorc.orc:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc.orc:
+ Use init functions for Orc code
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init), (gst_base_audio_src_init),
- (gst_base_audio_src_provide_clock),
- (gst_base_audio_src_set_property),
- (gst_base_audio_src_get_property), (gst_base_audio_src_create):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- Allow othe clocks than the internal clock to be used for the pipeline.
- Add property to disable clock provide.
- API: GstBaseAudioSrc::provide-clock
+2010-08-26 15:17:20 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-09-10 Stefan Kost <ensonic@users.sf.net>
+ * gst/volume/gstvolume.c:
+ volume: make the orc usage for float conditional again
+ See bug #628009. The tests still fail in the orc code (which we just don't call
+ now).
- * gst/playback/gstdecodebin2.c:
- Don't leak request pads. Fixes #475395.
+2010-08-25 12:19:05 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add support for 2vuy
+ It is the apple alternative for Microsofts UYVY.
+ (http://ntta.szm.com/Tutors/FourCC.htm)
+ Only use the UYVY for the caps to enable support in other
+ gstreamer elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=627924
- Patch by: René Stadler <mail at renestadler dot de>
+2010-08-25 19:01:57 +0300 Stefan Kost <ensonic@users.sf.net>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximage_buffer_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_class_init):
- Correctly chain up finalize with the parent class to prevent
- memory leaks. Fixes #474880.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ * gst/volume/gstvolumeorc.orc:
+ volume: enable ORC for float in volume
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+2010-08-25 11:19:31 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_choose_func):
- * tests/check/elements/volume.c: (GST_START_TEST):
- Revert the latest change: floating point samples are allowed to
- have any value, not only values in the range [-1,1]. Thanks to Andy
- Wingo for noticing.
- Also fix processing of int32 samples with volumes > 4 by making the
- unity value smaller which prevents overflows.
+ * configure.ac:
+ * gst-libs/gst/tag/gstexiftag.c:
+ configure: Add check for log2
+ Adds check for log2 and only use it in exif library if it is
+ available.
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-25 15:32:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- * tests/check/libs/rtp.c:
- Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Link to $(LIBM) for pow(), log2() and friends
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-25 08:41:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Move some tags to their correct IFDs
+ Put some tags in their correct IFDs
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Fix up GstRTPHeader helper struct so that compilers will not under
- any circumstances add padding in between our fields, as currently
- happens with MSVC on win32, because that would lead to us sending
- out RTP payloads with broken RTP headers (#471194).
- Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
-
- * tests/check/Makefile.am:
- * tests/check/libs/.cvsignore:
- * tests/check/libs/rtp.c:
- Add some simple unit tests for GstRTPBuffer. Some are disabled
- because the code tested still needs fixing (set_csrc() does not work).
+2010-08-20 16:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Always write FlashPixVersion tag
+ FlashPixVersion is mandatory and constant. Write it always.
- * win32/MANIFEST:
- * win32/common/gstrtsp-enumtypes.c:
- * win32/common/gstrtsp-enumtypes.h:
- * win32/common/interfaces-enumtypes.c:
- * win32/common/interfaces-enumtypes.h:
- * win32/common/multichannel-enumtypes.c:
- Add rtsp enumtypes (#474384) and update others.
+2010-08-20 15:59:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-06 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds flash tags mapping
+ Adds a mapping for GST_TAG_CAPTURING_FLASH_FIRED/_MODE to
+ the exif Flash tag.
+ Tests included.
- * configure.ac:
- Fix configure check for HAVE_LIBXML_HTML.
+2010-08-19 15:47:18 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ * win32/common/libgsttag.def:
+ tag: exif: More photography mappings
+ Adds mappings from:
+ GST_TAG_CAPTURING_EXPOSURE_PROGRAM -> ExposureProgram
+ GST_TAG_CAPTURING_EXPOSURE_MODE -> ExposureMode
+ GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE -> SceneCaptureType
+ GST_TAG_CAPTURING_GAIN_ADJUSTMENT -> GainControl
+ GST_TAG_CAPTURING_WHITE_BALANCE -> WhiteBalance
+ GST_TAG_CAPTURING_CONTRAST -> Constrast
+ GST_TAG_CAPTURING_SATURATION -> Saturation
+ Also renames gst_tag_image_orientation_from_exif_value and
+ gst_tag_image_orientation_to_exif_value to remove the 'gst'
+ prefix and not including in the win32 defs.
+ Tests included.
+
+2010-08-19 09:39:39 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Add macros for the exif ids
+ Use macros for exif ids to avoid having those numbers spread
+ all over the code.
+
+2010-08-17 15:56:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds photography tags mappings
+ Adds the following mappings for the exif helper:
+ * GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO -> DigitalZoomRatio
+ * GST_TAG_CAPTURING_FOCAL_LENGTH -> FocalLength
+ * GST_TAG_CAPTURING_SHUTTER_SPEED -> ExposureTime, ShutterSpeedValue
+ * GST_TAG_CAPTURING_FOCAL_RATIO -> FNumber, ApertureValue
+ * GST_TAG_CAPTURING_ISO_SPEED -> ISOSpeed, PhotographicSensitivity
+ Tests included.
+
+2010-08-17 15:05:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds mapping for GST_TAG_APPLICATION_DATA
+ Adds mapping for GST_TAG_APPLICATION_DATA to the exif
+ 'maker-note' tag.
- * tests/check/libs/.cvsignore:
- Ignore more, in case the build bots work again one day.
+2010-08-20 14:54:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: Adds GST_TAG_CAPTURE_FLASH_FIRED/_MODE
+ Adds a new tag for informing if flash was used while
+ capturing an image and the flash mode selected by the
+ user during this capture
+ API: GST_TAG_CAPTURING_FLASH_FIRED
+ API: GST_TAG_CAPTURING_FLASH_MODE
+ https://bugzilla.gnome.org/show_bug.cgi?id=626651
- Reviewed by: Stefan Kost <ensonic@users.sf.net>
+2010-08-17 07:21:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/fft/Makefile.am:
- * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
- * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
- * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
- * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
- * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
- * gst-libs/gst/fft/gstfft.h:
- * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
- (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
- (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
- * gst-libs/gst/fft/gstfftf32.h:
- * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
- (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
- (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
- * gst-libs/gst/fft/gstfftf64.h:
- * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
- (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
- (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
- * gst-libs/gst/fft/gstffts16.h:
- * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
- (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
- (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
- * gst-libs/gst/fft/gstffts32.h:
- * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
- (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
- (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
- (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
- * gst-libs/gst/fft/kiss_fft_f32.h:
- * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
- (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
- (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
- (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
- * gst-libs/gst/fft/kiss_fft_f64.h:
- * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
- (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
- (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
- (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
- * gst-libs/gst/fft/kiss_fft_s16.h:
- * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
- (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
- (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
- (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
- * gst-libs/gst/fft/kiss_fft_s32.h:
- * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
- (kiss_fftr_f32), (kiss_fftri_f32):
- * gst-libs/gst/fft/kiss_fftr_f32.h:
- * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
- (kiss_fftr_f64), (kiss_fftri_f64):
- * gst-libs/gst/fft/kiss_fftr_f64.h:
- * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
- (kiss_fftr_s16), (kiss_fftri_s16):
- * gst-libs/gst/fft/kiss_fftr_s16.h:
- * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
- (kiss_fftr_s32), (kiss_fftri_s32):
- * gst-libs/gst/fft/kiss_fftr_s32.h:
- * gst-libs/gst/fft/kiss_version:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Add libgstfft, a FFT library based on Kiss FFT which is
- BSD licensed. Supported sample formats are int16, int32,
- float and double. For those formats a real FFT and IFFT
- can be done, different windowing functions can be applied
- and functions for extracting the magnitude and phase exist.
- Fixes #468619.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: More photography related tags
+ API: GST_TAG_CAPTURING_EXPOSURE_PROGRAM
+ API: GST_TAG_CAPTURING_EXPOSURE_MODE
+ API: GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE
+ API: GST_TAG_CAPTURING_GAIN_ADJUSTMENT
+ API: GST_TAG_CAPTURING_WHITE_BALANCE
+ API: GST_TAG_CAPTURING_CONTRAST
+ API: GST_TAG_CAPTURING_SATURATION
+ Fixes #626651
+
+2010-08-17 06:47:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
- Integrate libgstfft into the docs.
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ tag: Adds some basic photography tags
+ Adds the following basic photography tags.
+ API: GST_TAG_CAPTURING_SHUTTER_SPEED
+ API: GST_TAG_CAPTURING_FOCAL_RATIO
+ API: GST_TAG_CAPTURING_FOCAL_LENGTH
+ API: GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO
+ API: GST_TAG_CAPTURING_ISO_SPEED
+ Fixes #626651
- * tests/check/Makefile.am:
- * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
- Add unit tests for libgstfft, currently only testing the FFT.
- Unit tests for IFFT will follow soon.
-
-2007-09-05 Wim Taymans <wim.taymans@gmail.com>
-
- Patch by: Peter Kjellerstedt <pkj at axis com>
-
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
- (gst_sdp_message_init), (gst_sdp_message_uninit),
- (is_multicast_address), (gst_sdp_message_as_text),
- (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
- (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
- (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
- (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
- (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
- (gst_sdp_media_init), (gst_sdp_media_uninit),
- (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
- (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
- (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
- (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
- (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Separate INIT_ARRAY() and related macros into two versions, one for
- structures and one for pointers (e.g., INIT_ARRAY() and
- INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
- lists of emails and phone numbers.
- Add missing const as appropriate.
- Change all gint to guint since they all actually represent unsigned
- values.
- Do not use time as a variable name as it shadows the global time().
- Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
- Actually implement gst_sdp_message_add_time().
- Make gst_sdp_message_add_time() take repeat times as an argument.
- Store repeat times in GstSDPTime as a GArray rather than as gchar**.
- Corrected the definition of gst_sdp_media_get_bandwidth() (was
- misspelled as badwidth).
- gst-indented and a little clean up. Fixes #471067.
-
-2007-09-05 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/volume/gstvolume.c: (volume_choose_func),
- (volume_process_double), (volume_process_double_clamp),
- (volume_process_float_clamp):
- Correctly clamp float/double samples in the [-1.0,1.0] range to
- prevent weird effects.
- * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
- Add unit tests for all samples types that had none before.
-
-2007-09-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-24 15:06:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Need to include stdlib.h for abs() here too.
+ * configure.ac:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ audioconvert: Require ORC 0.4.7 for the loadl/storel opcodes
+ And update disted files to allow compilation with no or too old ORC.
-2007-09-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-24 11:39:09 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
- * gst/playback/gststreaminfo.c:
- Fix build.
+ * gst/adder/gstadder.c:
+ adder: Make sure FLUSH_STOP is always sent after a flushing seek.
+ Send FLUSH_STOP right after forwarding the seek event upstream if necessary.
+ This makes sure that adder->srcpad is not left flushing if seeking fails or if
+ upstream is blocked.
+ The same fix was already applied to videomixer in 49b2a946.
-2007-09-05 Stefan Kost <ensonic@users.sf.net>
+2010-08-24 11:11:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gststreaminfo.c:
- Clean up some half-disabled code and comment.
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ audioconvert: Use ORC for the float<->int32 conversion
+ This should speed up standard Vorbis encoding and decoding pipelines a bit.
+ Thanks to David Schleef for the assistance to get the ORC code right
+ and explaining everything.
-2007-09-04 Wim Taymans <wim.taymans@gmail.com>
+2010-08-24 10:12:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Support NV21 too and minor cleanups
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_payload_audio_handle_event):
- Return FALSE from the event handler to let the parent class handle the
- event. Fixes #446766.
+2010-08-24 10:03:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
- Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
+ * gst-libs/gst/video/video.c:
+ video: Fix component width for NV12/NV21
+ Both have width/2 as component width for the chroma planes.
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Bump the MTU to 1400.
+2010-08-24 09:51:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-03 Johan Dahlin <jdahlin@async.com.br>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix NV21 rendering
+ Using the same as for NV12 will result in wrong colors and crashes.
- * gst/typefind/gsttypefindfunctions.c (plugin_init):
- Add an audio/x-nsf typefind function for the nsfdec element.
+2010-08-23 18:51:18 -0400 Chris Shoemaker <chris.shoemaker@cox.net>
-2007-09-03 Renato Filho <renato.filho@indt.org.br>
- * gst/playback/gstplaybasebin.c:
- Included "myth://" on stream_uris list for enable buffering to mythtv files
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gstclockoverlay.h:
+ clockoverlay: only rerender text if time string has changed
+ The textoverlay element will rerender the text string whenever
+ overlay sets the 'need_render' flag to TRUE. Previously, we
+ lazily set the flag to TRUE every time the time string was requested.
+ Now, we save a copy of the previously given string, and only set
+ 'need_render' to TRUE if the string has changed.
+ In my tests with a 30fps video stream, and a time string including
+ a seconds field, this change reduced the CPU usage of the clockoverlay
+ element from 60% to 5%.
+ Fixes bug #627780.
+
+2010-08-23 13:59:38 -0500 Rob Clark <rob@ti.com>
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: add NV12 support
+ Fixes bug #627768.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
- (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
- (gst_rtcp_unix_to_ntp):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Fix parsing of RB blocks.
- Fix docs.
- Added helper functions to convert to/from UNIX and NTP time.
- API: gst_rtcp_ntp_to_unix()
- API: gst_rtcp_unix_to_ntp()
-
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
- (gst_rtp_buffer_get_header_len),
- (gst_rtp_buffer_get_extension_data),
- (gst_rtp_buffer_get_payload_subbuffer),
- (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
- (gst_rtp_buffer_ext_timestamp):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Fix some more docs.
- Implement handling of packets with extensions.
- Fix padding check in _validate().
- Added function to get extension data.
- API: gst_rtp_buffer_get_header_len()
- API: gst_rtp_buffer_get_extension_data()
+2010-08-20 12:03:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Mark duplicated frames with the GAP flag
+ We currently don't use the GAP flag for video and the docs say
+ that this is for buffers, that have been created to fill a gap
+ and contains neutral data. For video this is the previous frame.
+ This information can be used by encoders to encode the duplicated
+ frames more efficiently. See bug #627459.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init),
- (gst_base_rtp_depayload_set_gst_timestamp):
- Add some more docs for the queue-delay property and fix a typo in a
- comment.
+2010-08-19 18:51:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- Fix typo.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Add back support for 8 bit paletted RGB
+ This was removed by 3a00a97fd2b4015e93cdcabaa75da406aa599570
+ while making the pad template caps more compact.
+ Fixes bug #626629.
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+2010-08-18 16:45:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- When skew slaving, try to hover around the middle of a segment so that
- we at most drift by half a segment.
- If we are aligning in the oposite direction of the clock skew, we don't
- have to resync.
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsasrc/sink: add property to get the card name
+ fixes #627203
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+2010-08-18 16:42:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_set_gst_timestamp):
- Be less silly with the segment start, just apply the clock-base to the
- timestamp.
+ * ext/alsa/gstalsa.c:
+ * ext/alsa/gstalsa.h:
+ alsa: add method to retrieve the card name
+ Reuse an existing method to retrieve the card name.
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+2010-08-18 12:34:07 +0200 American Dynamics <GStreamer-Bugs at tycosp.com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init),
- (gst_base_rtp_depayload_finalize),
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_handle_sink_event),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_change_state):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Deprecate the queue handling thread thing and remove the code.
- Use new method to calculate the extended timestamp.
-
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
-
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_sdes_copy_entry):
- Use g_strndup which does exactly what we want.
-
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
- (gst_rtp_buffer_ext_timestamp):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Add helper function to compare seqnums.
- Add helper function to calculate extended timestamps.
- API: gst_rtp_buffer_compare_seqnum()
- API: gst_rtp_buffer_ext_timestamp()
+ basertpdepay: don't clear the discont flag too early
+ Set the discont flag when we receive a DISCONT buffer and only clear the discont
+ state when we pushed out a DISCONT buffer.
+ Fixes #626869
-2007-08-30 Wim Taymans <wim.taymans@gmail.com>
+2010-08-14 19:08:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_sdes_get_entry),
- (gst_rtcp_packet_sdes_copy_entry):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Fix and document SDES item data function.
- Add new function that makes a proper copy of SDES item data.
- API: gst_rtcp_packet_sdes_copy_entry()
+ * gst-libs/gst/app/gstappsink.c:
+ docs: fix typo in appsink docs so function gets cross-referenced properly
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
+2010-08-14 19:02:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * common:
* configure.ac:
- * gst/Makefile.am:
- The tcp and subparse plugins are under gst, but not totaly free of
- dependencies. Handle selection inconfigure.ac, so that they show up
- on the final list of what is build and what is not. Maybe they should
- better be moved to ext.
-
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
-
- Patch by: Daniel Díaz <yosoy@danieldiaz.org>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ build: use new AG_GST_PKG_CONFIG_PATH m4 macro from common
+ Sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
+ (avoids trailing ':' in PKG_CONFIG_PATH used).
- * configure.ac:
- * gst/Makefile.am:
- Check if libxml provides HTML parser which subparse needs.
- Fixes #451970.
+2010-08-14 18:36:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: set PKG_CONFIG_PATH so that our in-tree libs come first when calling scanner
+ When calling gobject-introspection scanner, make sure our own
+ freshly-built libs within the source tree (well, build dir) come
+ first in the PKG_CONFIG_PATH. May or may not help to make sure
+ that it doesn't pick up older external plugins-base libs (or
+ .gir files) from outside the source tree / build directory as
+ dependencies of the introspected lib instead of using the
+ stuff we just built in a sibling directory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=623698
+
+2010-08-06 17:16:27 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/alsa/gstalsa.c:
- Fix typo and compilation on big endian systems.
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playback: Delay usage of GstFactoryList
+ By delaying it to when it's actually needed, we speed things up a bit
+ since some elements might have been added/removed in between.
+ https://bugzilla.gnome.org/show_bug.cgi?id=626718
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-17 09:10:11 +0200 Robert Swain <robert.swain@collabora.co.uk>
- * gst/subparse/gstssaparse.c:
- Convert SSA newline codes into actual newline characters (#470766).
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2: uridecodebin: add property to configure ring buffer size
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-13 17:23:46 +0300 Stefan Kost <ensonic@users.sf.net>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/pbutils/install-plugins.c:
- * gst-libs/gst/pbutils/install-plugins.h:
- * tests/check/libs/pbutils.c:
- API: also add gst_install_plugins_supported() while we're at it
- (see #470456).
+ * common:
+ Automatic update of common submodule
+ From 3e8db1d to ec60217
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-13 13:59:08 +0300 Stefan Kost <ensonic@users.sf.net>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/pbutils/missing-plugins.c:
- * gst-libs/gst/pbutils/missing-plugins.h:
- * tests/check/libs/pbutils.c:
- API: add gst_missing_*_installer_detail_new() convenience API so
- that applications that know exactly what they're missing can request
- installer detail strings for those items directly instead of having
- to first create a dummy missing-plugin message and then get the
- installer detail string from that. Fixes #470456.
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ plugin-docs: the <TITLE> tag should come right after <FILE>.
+ Fixes missing plugin entries. If the object name, e.g. GstGIOSrc came before the
+ title, we ended up with differnt section_id in the generated docbook.
-2007-08-27 Jan Schmidt <thaytan@mad.scientist.com>
+2010-08-12 18:14:38 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- We need to set up delayed-linking whenever the caps are non-fixed,
- not just when there are multiple types - use gst_pad_is_fixed()
- to test.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix the classification.
+ Change "Src" into "Source" (we use that elsewhere). I did not keept "Src" as it
+ is quite unlikely that someone plugs appsrc by searching the registry by classification.
-2007-08-26 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-12 15:26:08 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_plugin_message_get_installer_detail):
- Add missing separator in PID fallback case.
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: remove 'dir' out variable
+ Alsa seems to expect that we initialize it. Remove the variable and pass NULL
+ as we actually don't use it. In alsasink also #ifdef one section that is
+ grabing diagnostics to be disabled, when logging is disabled (the code was
+ using the out parameter as well).
+ Fixes #626125
-2007-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+2010-08-12 11:46:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/Makefile.am:
- There is no GST_PLUGINS_BASE_LIBS defined.
-
- * ext/alsa/gstalsa.c:
- * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
- Add support for ALSA 24-bit formats.
- snd_pcm_delay can return an error code, especially
- during XRUNS. In that case, the best we can do is assume
- delay = 0.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: remove chroma-site and color-matrix fields from RGB caps
- * gst/audioconvert/Makefile.am:
- Add flags from -base before any more-remote dependencies.
+2010-08-11 12:49:40 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: prevent deadlock with _chain when deactivating pad
+ Fixes #626581.
- Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
+2010-08-12 12:50:27 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/volume/gstvolume.c: (volume_choose_func),
- (volume_update_real_volume), (gst_volume_set_volume),
- (gst_volume_init), (volume_process_int32),
- (volume_process_int32_clamp), (volume_process_int24),
- (volume_process_int24_clamp), (volume_process_int16),
- (volume_process_int16_clamp), (volume_process_int8),
- (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
- * gst/volume/gstvolume.h:
- Add support for int32, int24 and int8 to the volume element.
- Fixes #445529.
+ * gst/playback/Makefile.am:
+ playback: bad bad editor, readd missing line to fix the build
-2007-08-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-12 12:08:35 +0300 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
* tests/examples/Makefile.am:
- Fix even more.
+ * tests/examples/playback/.gitignore:
+ * tests/examples/playback/Makefile.am:
+ * tests/examples/playback/decodetest.c:
+ * tests/examples/playback/test.c:
+ * tests/examples/playback/test2.c:
+ * tests/examples/playback/test3.c:
+ * tests/examples/playback/test4.c:
+ * tests/examples/playback/test5.c:
+ * tests/examples/playback/test6.c:
+ * tests/examples/playback/test7.c:
+ * tests/icles/Makefile.am:
+ * tests/icles/playback/.gitignore:
+ * tests/icles/playback/Makefile.am:
+ * tests/icles/playback/decodetest.c:
+ * tests/icles/playback/test.c:
+ * tests/icles/playback/test2.c:
+ * tests/icles/playback/test3.c:
+ * tests/icles/playback/test4.c:
+ * tests/icles/playback/test5.c:
+ * tests/icles/playback/test6.c:
+ * tests/icles/playback/test7.c:
+ tests/playback: due to popular demand mv them from examples to icles
+ The tests are toys and not reference demos.
+
+2010-08-12 10:02:56 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: send preroll buffer when delaying preroll eos
+ That is, if eos is received which will not be forwarded, and the stream
+ has not yet seen any data, then send a buffer to preroll downstream
+ (which might otherwise be accomplished by the eos event).
+
+2010-08-12 10:01:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaysink.c:
+ playsink: remove some heuristic in chain configuration code
+ .. since queues are now inserted unconditionally.
- * configure.ac:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * ext/gnomevfs/gstgnomevfssrc.c:
- * ext/gnomevfs/gstgnomevfssrc.h:
- * gst-libs/gst/Makefile.am:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- * sys/v4l/v4lsrc_calls.c:
- * tests/examples/Makefile.am:
- * win32/common/config.h:
- Revert unwanted commit. many thanks to moap. I want a fix for
- https://thomas.apestaart.org/moap/trac/ticket/239
+2010-08-11 10:27:35 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: update subtitle handling for streamsynchronizer
+ Streamsynchronizer excepts to see stream-changed msg for all streams, but to
+ arrange for this, video and subtitle streams need to be decoupled by means
+ of queues (due to pad blocks that may occur).
+ Fixes #626463.
- * gst/volume/gstvolume.c:
- Move passthrough below gst_object_sync_values(). Fixes #442654.
+2010-08-10 13:06:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-22 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaysink.c:
+ playsink: always have a queue in chain head to aid streamsynchronizer
+ Specifically, as the latter may have one thread pushing EOS to several streams,
+ that needs to be decoupled into various thread to prevent preroll hanging
+ problems.
- * gst-libs/gst/audio/audio.c:
- Clarify the docs a little.
+2010-08-10 11:28:43 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-22 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: drop lock when pushing eos downstream
+ ... to prevent deadlock (e.g. upon seek) when downstream waits in preroll.
- * gst/volume/gstvolume.c:
- Enable liboil for float and add more details about problems with
- int16.
+2010-08-10 11:19:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: clear stream eos state on FLUSH and new stream
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
- Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
+2010-08-10 11:19:22 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaysink.c:
+ playsink: set READY sinks to NULL before freeing chain upon failure
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
- When calculating the first timestamp of the buffers, don't go below 0
- and clip the samples because the offset was on the eos page.
- Fixes #466717.
+2010-08-12 10:49:59 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+ * configure.ac:
+ * gst/playback/.gitignore:
+ * gst/playback/Makefile.am:
+ * gst/playback/decodetest.c:
+ * gst/playback/test.c:
+ * gst/playback/test2.c:
+ * gst/playback/test3.c:
+ * gst/playback/test4.c:
+ * gst/playback/test5.c:
+ * gst/playback/test6.c:
+ * gst/playback/test7.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/playback/.gitignore:
+ * tests/examples/playback/Makefile.am:
+ * tests/examples/playback/decodetest.c:
+ * tests/examples/playback/test.c:
+ * tests/examples/playback/test2.c:
+ * tests/examples/playback/test3.c:
+ * tests/examples/playback/test4.c:
+ * tests/examples/playback/test5.c:
+ * tests/examples/playback/test6.c:
+ * tests/examples/playback/test7.c:
+ playback: move tests from plugin-dir to tests/examples/playback
+
+2010-08-11 18:08:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
- (gst_ogg_demux_collect_chain_info):
- Also submit the eos page when trying to find the first timestamp.
- See #466717.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: Suggest caps with different width/height if bufferalloc is called with impossible width/height
-2007-08-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-08-11 17:16:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/audio.h:
- Use gst_util_uint64_scale() instead of doing the math
- with double for GST_FRAMES_TO_CLOCK_TIME() and
- GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
- prevents rounding errors. Fixes #467667.
+ * tests/check/elements/videoscale.c:
+ videoscale: Add some debug output to the videoscale negotiation test
-2007-08-17 Wim Taymans <wim.taymans@gmail.com>
+2010-08-11 17:03:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
- (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Small cleanups.
- On shutdown, don't read the control socket yet.
- Set timeout value correctly in all cases.
- Add function to check if the server accepts reads or writes.
- API: gst_rtsp_connection_poll()
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Only set the PAR if the caps already had a PAR
+ Otherwise we're producing different caps and basetransform thinks that it
+ can't passthrough buffer allocations, etc.
+ In 0.11 all video caps really should have the PAR set...
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- Fix compilation with -pedantic.
- Add enum for _poll.
+2010-08-11 17:00:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: It's not a bad thing if the preferred video format needs less bytes per frame
- Patch by: Olivier Crete <tester at tester ca>
+2010-08-11 08:47:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_getcaps):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Add getcaps vfunc to basertppayload. See #465146.
+ * gst-libs/gst/tag/gstexiftag.c:
+ tags: exif: Fix bug in inner ifd parsing
+ Do not use the result of inner ifd's parsing to increment
+ the current tag index. The reasons are:
+ 1) The function returns a boolean.
+ 2) The inner ifd's tags are in a separate table, so they shouldn't
+ interfere with its parent ifd table parsing.
-2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+2010-08-11 08:03:44 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
- Only post buffering messages when we are a stream.
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Put ExifVersion in the correct IFD
+ ExifVersion is from the 'exif' ifd, not the 0th ifd.
-2007-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-10 19:50:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/pbutils/install-plugins.c:
- * gst-libs/gst/pbutils/missing-plugins.c:
- Small docs fix and addition.
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Refactor functions declaration
+ Use some macros to declare serialization/deserialization
+ functions prototypes.
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-10 19:30:11 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-textoverlay.c:
- Add a dumb little test for textoverlay alignments.
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Add another DateTime mapping
+ datetimes can also be represented by the 0x132 tag. Map it, too.
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-10 11:29:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Dan Williams <dcbw redhat com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Fix bug on image-orientation parsing
+ Do not skip one extra tag when parsing image-orientation tags.
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextoverlay.h:
- API: add "line-alignment" property (#459334). Add gtk-doc blurb for
- "silent" property so there's a Since tag in the API reference.
+2010-08-10 10:57:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-08-12 Wim Taymans <wim.taymans@gmail.com>
+ * common:
+ Automatic update of common submodule
+ From bd2054b to 3e8db1d
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Improve caps negotiation so that downstream elements can confiure
- certain RTP properties by fixing them on the caps. See #465146.
- Add docs.
+2010-08-10 11:52:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ exiftag: Compare with G_MAXUINT16 instead of -1
+ Fixes a compiler warning on the OS X buildbot.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Mark as deprecated some macros which were presumably meant to be
- private API and accidentally exposed in the public header file.
- Also actually _init() lock (only works at the moment because the
- struct is zeroed out when created and the initial values in the
- mutex struct are zeroes too). (#459585)
+2010-08-09 18:04:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-08-10 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ xmp: exif: Adds GST_TAG_APPLICATION_NAME mappings
+ adds xmp and exif helper library mappings for GST_TAG_APPLICATION_NAME
+ tag.
- * docs/libs/Makefile.am:
- Remove cruft and do some cleanups.
+2010-08-04 13:01:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- Prepare for comming gtkdoc features (rebase against online docs).
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Write ExifVersion tag
+ Write ExifVersion tag unconditionally when creating exif
+ buffers. Might help other applications parsing of this data.
-2007-08-10 Michael Smith <msmith@fluendo.com>
+2010-08-04 13:02:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Debug output fixes.
- * tests/check/elements/audiorate.c: (do_perfect_stream_test),
- (GST_START_TEST):
- Change the number of buffers used; 500 is too many and leads to
- timeouts.
+ * tests/check/libs/tag.c:
+ tests: tag: Test to try to serialize multiple exif tags
+ Adds a new test for exif data that tries serializing data
+ from multiple ifd tables and check if it works.
-2007-08-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-08-09 17:25:07 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstqueue2.c:
- * gst/videorate/gstvideorate.c:
- Printf format fixes (#465028).
+ * gst-libs/gst/tag/gstexiftag.c:
+ tags: exif: Fix inner tags offset rewriting
+ Fixes a bug that made exif helper lib fail to rewrite inner ifd tags
+ offsets when there were more than 1 inner ifd.
-2007-08-09 Michael Smith <msmith@fluendo.com>
+2010-07-22 17:29:57 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- If we have a large (> 1 second) discontinuity, push a series of
- smaller buffers rather than a single very large buffer. Avoids
- unreasonably large single buffer allocations when encountering a
- large gap.
- * tests/check/elements/audiorate.c: (GST_START_TEST),
- (audiorate_suite):
- Add a test for this.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: configurable text color and position
+ Rather than only left, right, top, etc, allow for horizontal and vertical
+ positioning on a scale from 0 to 1.
+ Also cater for configuring rendered text color.
+ Fixes #624920.
+ API: GstTextOverlay:xpos
+ API: GstTextOverlay:ypos
+ API: GstTextOverlay:color
-2007-08-09 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-21 14:20:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_commit),
- (queue_remove_probe), (queue_threshold_reached):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add solid-color pattern
+ ... which generalizes the current listing of white, black, etc.
+ In particular, also allow specifying alpha channel, and modify
+ some structures and pattern filling to cater for alpha value as well.
+ Fixes #624919.
+ API: GstVideoTestSrc:solid-color
- Patch by: Josep Torra Valles <josep@fluendo.com>
- Fixes: #465015
- Make sure we remove the check_queues buffer probe from the
- correct queue to avoid racily going back to "buffering 99%" when
- buffering is actually complete.
+2010-08-08 17:42:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Also, fix the spelling of Josep's surname in the ChangeLog.
+ * ext/ogg/gstoggstream.c:
+ oggstream: static forward declarations are forbidden by the C standard
+ ...and actually cause compiler errors on VC++. Change it to an extern
+ forward declaration and non-static definition.
-2007-08-09 Stefan Kost <ensonic@users.sf.net>
+2010-08-05 13:56:29 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggmux.c:
- Do not leak oggmux instance.
-
- * ext/vorbis/vorbisenc.c:
- Also log values.
+ * common:
+ Automatic update of common submodule
+ From 2004d03 to bd2054b
-2007-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-08-04 19:24:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/hu.po:
- * po/it.po:
- * po/nl.po:
- * po/uk.po:
- * po/vi.po:
- Updated translations.
+ * configure.ac:
+ configure: Check if the compiler supports ISO C89 or C99 and which parameters are required
+ This first checks what is required for ISO C99 support and sets the relevant
+ compiler parameters and if no C99 compiler is found, it checks for a
+ C89 compiler. This enables us to check for and use C89/C99 functions
+ that gcc hides from us without the correct compiler parameters.
-2007-08-08 Stefan Kost <ensonic@users.sf.net>
+2010-08-04 15:18:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- patch by: Yang Hong <hongyang@redflag-linux.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosink/baseaudiosrc: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY
+ Otherwise the clocks are redistributed every time the pipeline
+ goes to PAUSED, which is quite expensive.
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextoverlay.h:
- Add 'silent' property to GstTimeOverlay. Fixes #462979
+2010-08-03 15:03:27 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-08-08 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Make xmp lib aware for the different tag types
+ Makes the xmp helper lib aware that the tags can be simple,
+ sequences or bags (there is still struct and alt, but those
+ aren't handled yet). Adding this info makes serialization
+ and deserialization more consistent.
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2010-08-02 09:56:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins.args:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
- (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
- (gst_uri_decode_bin_get_property), (gen_source_element):
- Add connection-speed property. Fixes #464690.
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmp: Add a new layer of indirection (GstXmpSchema)
+ Instead of storing all tags in a single hashtable, store them
+ grouped by schema in a GstXmpSchema, and add those to the toplevel
+ hashtable.
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+2010-08-03 14:37:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Make bag tags deserialization work correctly
+ If we find a bag of tags of type string in the xmp packet, we
+ should concat them, this is not the ideal approach, but at
+ least works for now as we don't know what type of tag it
+ is (simple, structure, seq, alt or bag)
- * configure.ac:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect):
- Fix compilation on windows. Fixes #464320.
+2010-08-04 21:44:22 +1000 Jan Schmidt <thaytan@noraisin.net>
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+ * tests/examples/seek/seek.c:
+ examples/seek: Don't unpause on clock-lost unless playing
+ If the pipeline is paused by the user, don't pause/unpause
+ on clock-lost.
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2010-07-02 12:10:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (queue_threshold_reached),
- (gen_source_element), (setup_substreams),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (gst_play_base_bin_get_streaminfo_value_array):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (gst_play_bin_handle_redirect_message):
- Move connection-speed property from playbin to playbasebin so that we
- can also configure it in source elements that have the connection-speed
- property. Fixes #464028.
- Add some debug info here and there.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: improve debugging
-2007-08-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-07-02 12:09:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
- Properly respond to conversion queries. Fixes #464079.
+ * gst-libs/gst/audio/gstringbuffer.h:
+ ringbuffer: whitespace fixes
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-28 10:53:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
- (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
- (gst_audio_test_src_init_sine_table),
- (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Add float/double and int32 support to audiotestsrc. Fixes #460422.
- Also set the default volume to the default value specified in the
- GParamSpec.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: use more efficient way of getting caps
+ When inspecting the caps of a pad, try to get the pad _CAPS first before calling
+ the getcaps function.
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-08-02 11:06:00 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Jens Granseuer <jensgr at gmx dot net>
+ * tests/check/pipelines/oggmux.c:
+ oggmux: Fix test build when theora and vorbis aren't available
+ Ifdef properly to avoid build failures
- * gst/audioconvert/gstaudioquantize.c:
- Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
+2010-08-01 06:50:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-08-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Detect avc1 ftyp as video/quicktime
+ Detects avc1 ftyp as video/quicktime (iso variant)
- * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
- Add rdt manager for rdt transport.
- Fix parsing of RDT transport.
+2010-07-27 11:25:12 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: removing useless locking
+ Everything in the xmp helper lib is initiallized once and on a thread
+ safe way, and after that there are only reads going on, no more
+ writing. Based on that, drop the locking.
- * configure.ac:
- Back to CVS
+2010-06-20 23:53:38 +1000 Jan Schmidt <thaytan@noraisin.net>
-=== release 0.10.14 ===
+ * tests/examples/seek/jsseek.c:
+ jsseek: Set joystick io encoding to 'NULL'
+ Fix problems with newer glib reporting bad encodings on the binary
+ data emerging from the joystick device fd.
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-26 20:25:55 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- releasing 0.10.14, "Light Years Ahead"
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: fix printf format compiler warnings
+ Make OSX build bot happy.
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-26 18:23:33 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/libs/audio.c: (GST_START_TEST):
- Fix the test to reflect the behaviour of gst_audio_clip_buffer.
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: initialize datetime variable in xmp tag parsing code
+ Fixes (correct) compiler warning on the OSX build bot.
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-26 17:48:14 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- When clipping a buffer with no timestamp, assume it is
- within the segment without warnings.
+ * configure.ac:
+ configure: require core from git
+ For GstDateTime stuff used in libgsttag.
- Fixes: #460978
+2010-07-26 17:04:02 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-27 Wim Taymans <wim.taymans@gmail.com>
+ * .gitignore:
+ * configure.ac:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/gstpluginsbaseversion.c:
+ * gst-libs/gst/pbutils/gstpluginsbaseversion.h.in:
+ * gst-libs/gst/pbutils/pbutils.h:
+ * tests/check/libs/pbutils.c:
+ * win32/common/libgstpbutils.def:
+ pbutils: add compile time and runtime version checks for gst-plugins-base
+ So people can check what version of the gst-plugins-base libs they're
+ building against or linked against.
+ API: GST_PLUGINS_BASE_VERSION_MAJOR
+ API: GST_PLUGINS_BASE_VERSION_MINOR
+ API: GST_PLUGINS_BASE_VERSION_MICRO
+ API: GST_PLUGINS_BASE_VERSION_NANO
+ API: GST_CHECK_PLUGINS_BASE_VERSION
+ API: gst_plugins_base_version()
+ API: gst_plugins_base_version_string()
+
+2010-06-30 16:36:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Map GST_TAG_DATE_TIME
+ Adds mapping to the exif helper library for GST_TAG_DATE_TIME.
+ Tests included.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594504
- * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
- Fire the signal on the object, not the interface.
+2010-06-23 12:02:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Maps GST_TAG_DATE_TIME
+ Adds mapping for GST_TAG_DATE_TIME.
+ Tests included.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594504
- * gst-libs/gst/rtsp/.cvsignore:
- Ber. Don't include the full path, idiot.
+2010-07-26 16:05:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate the pixel-aspect-ratio if necessary
- * gst-libs/gst/rtsp/.cvsignore:
- Ignore generated files.
+2010-07-24 18:17:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Delay EOS events until all streams are EOS
+ This fixes a race condition in playbin2's gapless mode, where the
+ EOS of other streams might arrive in the sinks before the last stream
+ ends and the switch to the new track happens. The EOS sinks won't
+ accept any new data then and playback stops.
+ To prevent this, delay all EOS events until all streams are EOS
+ and advance the sinks of the EOS streams by filler newsegment
+ events if necessary.
+ Fixes bug #625118.
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/interfaces-marshal.list:
- * gst-libs/gst/interfaces/rtspextension.c:
- * gst-libs/gst/interfaces/rtspextension.h:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtsp.h:
- * gst-libs/gst/rtsp/gstrtspextension.c:
- (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
- (gst_rtsp_extension_detect_server),
- (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
- (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
- (gst_rtsp_extension_configure_stream),
- (gst_rtsp_extension_get_transports),
- (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
- * gst-libs/gst/rtsp/gstrtspextension.h:
- * gst-libs/gst/rtsp/rtsp-marshal.list:
- Move the rtspextension.h interface into gstrtspextension.h
- as part of libgstrtsp instead of libgstinterfaces, because it's
- only for use within plugins, not applications.
- Add stuff to do the enum & marshal generation needed in libgstrtsp now.
- Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
- signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
- is abstract.
+2010-06-01 23:43:45 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: export 3gp profile in caps
+ This reads the 3gp profile from the major/compatible brands and puts
+ this as a 'profile' field in caps. This can be used by demuxers to
+ decide whether they can handle this stream or not. Also needed for
+ DLNA.
+ https://bugzilla.gnome.org/show_bug.cgi?id=620291
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/interfaces-marshal.list:
- * gst-libs/gst/interfaces/rtspextension.c:
- (gst_rtsp_extension_iface_init),
- (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
- * gst-libs/gst/interfaces/rtspextension.h:
- Fix marshaller for the send signal.
- Add URL to stream selection interface method.
+2010-07-24 11:48:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ * tests/icles/test-colorkey.c:
+ * tests/icles/test-xoverlay.c:
+ examples: Use cairo instead of to-be-deprecated GDK API
+ Fixes bug #625001.
- * gst-libs/gst/riff/Makefile.am:
- Pull in our dependencies from -base before those from outside.
+2010-07-24 09:22:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+ * common:
+ * configure.ac:
+ configure: set release date/time
+ Use the new AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO macro.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
- * gst-libs/gst/rtsp/gstrtspbase64.h:
- API: gst_rtsp_base64_decode_ip()
- Added function to decode Base64 in-place.
+2010-07-20 12:08:52 +0530 Parthasarathi Susarla <partha.susarla@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: detect enhanced AC-3
+ https://bugzilla.gnome.org/show_bug.cgi?id=623846
- * tests/check/libs/.cvsignore:
- Ignore the mixer test binary.
+2010-07-22 09:13:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gststreamsynchronizer.c:
+ * gst/playback/gststreamsynchronizer.h:
+ streamsynchronizer: Fix another deadlock when going PAUSED->READY while streams are waiting for the GCond
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
- Gratuitous comment change to trigger a rebuild on the buildbots.
+2010-07-20 21:05:45 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+ playsink: Switch to faster pad linking methods
+ Logic for choice of GST_PAD_LINK_CHECK_* is as follows:
+ * Where return of pad_link wasn't checked before : NOTHING
+ * Where linking is between known compatible elements : NOTHING
+ * All other cases : TEMPLATE_CAPS
+ Slashes down playsink reconfigure by up to 50% cpu time.
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
- (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
- (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
- (gst_sdp_media_get_format), (gst_sdp_media_get_information),
- (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
- (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
- (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
- (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
- (gst_sdp_media_get_attribute_val):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Constify args where we can.
+2010-07-19 15:58:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstscreenshot.c:
+ playsink: Set add-borders=true on the videoscale instances
+ This makes sure that we always keep the display aspect ratio and
+ add black borders if necessary, which is usually something you want
+ for viewing a video.
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/rtspextension.c:
- (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
- (gst_rtsp_extension_detect_server),
- (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
- (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
- (gst_rtsp_extension_configure_stream),
- (gst_rtsp_extension_get_transports),
- (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
- * gst-libs/gst/interfaces/rtspextension.h:
- Move interface for RTSP extensions from -good to here.
- Added helper methods to invoke interface methods.
-
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+2010-07-19 15:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
- (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
- (gst_rtsp_message_init_response),
- (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
- (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
- (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
- (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
- (gst_rtsp_message_get_body), (dump_key_value):
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse):
- * gst-libs/gst/rtsp/gstrtsprange.h:
- * gst-libs/gst/rtsp/gstrtsptransport.c:
- * gst-libs/gst/rtsp/gstrtspurl.c:
- Fix some more RTSP docs.
- Add some missing methods for dealing with messages.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Rename borders property to add-borders
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2010-07-19 09:39:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
- * gst-libs/gst/rtsp/gstrtspbase64.h:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (add_auth_header),
- (gst_rtsp_connection_write), (gst_rtsp_connection_send),
- (read_body), (gst_rtsp_connection_receive),
- (gst_rtsp_connection_next_timeout),
- (gst_rtsp_connection_reset_timeout),
- (gst_rtsp_connection_set_auth):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse):
- * gst-libs/gst/rtsp/gstrtspurl.h:
- Added beginnings of RTSP documentation.
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ videoscale: update disted orc files for latest changes
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2010-07-17 20:24:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/sdp/gstsdp.h:
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
- (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
- (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
- (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
- (gst_sdp_message_get_attribute_val),
- (gst_sdp_message_add_attribute), (gst_sdp_media_new),
- (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
- (gst_sdp_media_get_media), (gst_sdp_media_set_media),
- (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
- (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
- (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
- (gst_sdp_media_get_format), (gst_sdp_media_add_format),
- (gst_sdp_media_get_information), (gst_sdp_media_set_information),
- (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
- (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
- (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
- (gst_sdp_media_set_key), (gst_sdp_media_get_key),
- (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
- (gst_sdp_media_get_attribute_val_n),
- (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
- (print_media), (gst_sdp_message_dump):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Document the SDP library.
- Add some of the missing SDPMedia methods.
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_fill_borders.c:
+ * gst/videoscale/vs_fill_borders.h:
+ * gst/videoscale/vs_image.h:
+ videoscale: Add support for adding black borders to keep the DAR if necessary
+ Fixes bug #617506.
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2010-07-18 15:08:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
- * gst-libs/gst/rtsp/gstrtspbase64.h:
- * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
- (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
- (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
- (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
- (parse_response_status), (parse_request_line), (parse_line),
- (gst_rtsp_connection_read), (read_body),
- (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
- (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
- (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
- (gst_rtsp_connection_set_auth):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
- (gst_rtsp_strresult), (gst_rtsp_method_as_text),
- (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
- (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
- (gst_rtsp_find_method):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
- (gst_rtsp_message_new), (gst_rtsp_message_init),
- (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
- (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
- (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
- (gst_rtsp_message_free), (gst_rtsp_message_add_header),
- (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
- (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
- (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
- (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
- (gst_rtsp_message_dump):
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse), (gst_rtsp_range_free):
- * gst-libs/gst/rtsp/gstrtsprange.h:
- * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
- (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
- (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
- (range_as_text), (rtsp_transport_mode_as_text),
- (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
- (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
- (gst_rtsp_transport_free):
- * gst-libs/gst/rtsp/gstrtsptransport.h:
- * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
- (gst_rtsp_url_free), (gst_rtsp_url_set_port),
- (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
- * gst-libs/gst/rtsp/gstrtspurl.h:
- * gst-libs/gst/sdp/Makefile.am:
- * gst-libs/gst/sdp/gstsdp.h:
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
- (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
- (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
- (gst_sdp_attribute_init), (gst_sdp_message_new),
- (gst_sdp_message_init), (gst_sdp_message_uninit),
- (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
- (gst_sdp_media_uninit), (gst_sdp_media_free),
- (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
- (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
- (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
- (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
- (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
- (gst_sdp_message_get_attribute_val),
- (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
- (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
- (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
- (gst_sdp_media_get_attribute_val_n),
- (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
- (read_string), (read_string_del), (gst_sdp_parse_line),
- (gst_sdp_message_parse_buffer), (print_media),
- (gst_sdp_message_dump):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- Move SDP and RTSP from helper objects in -good to a reusable library.
- Use a proper gst_ namespace.
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix linear scaling of UYVY scanlines
+ Fixes bug #624656.
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+2010-07-17 19:57:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
- (vorbis_dec_flush_decode):
- Use the new buffer clipping function from gstaudio here.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix caps fixating if the height is fixed but the width isn't
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+2010-07-16 20:41:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
- * gst-libs/gst/audio/audio.h:
- * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
- API: Add buffer clipping function for raw audio buffers. Fixes #456656.
- Also add deprecation guards for gst_audio_structure_set_int() to the
- header.
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Remove interlaced scaling again
+ This behaviour was not preferred and caused visible image quality
+ degradations. The real solution would be, to apply a real
+ deinterlacing filter before scaling the frames.
+ Fixes bug #615471.
-2007-07-23 Stefan Kost <ensonic@users.sf.net>
+2010-07-16 19:06:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Cleanup the docs.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add helper method for filling the VSImage struct
-2007-07-23 Wim Taymans <wim.taymans@gmail.com>
+2010-07-18 11:43:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Dan Williams <dcbw at redhat dot com>
+ * tests/files/Makefile.am:
+ tests: don't forget to dist test file for typefinding unit test
- * gst/playback/gstplaybasebin.c:
- (gst_play_base_bin_get_streaminfo_value_array):
- Don't return NULL when querying the stream info value array but instead
- return an empty array. Fixes #459204.
+2010-07-18 11:38:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-23 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/gst/typefindfunctions.c:
+ * tests/files/623663.mts:
+ tests: add unit test for mpeg-ts typefinding bug
+ See #623663.
- * gst/playback/gsturidecodebin.c:
- Init debug category before using it.
+2010-07-18 11:24:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-21 Jan Schmidt <thaytan@noraisin.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: make mpeg-ts typefinder scan more data
+ We only look for packets with payload, but it appears there may be packets without,
+ which makes it harder to find the N packets with payload in a row that we need in
+ order to typefind this successfully, so scan some more data than necessary in the
+ optimistic scenario. Alternatively we could change IS_MPEGTS_HEADER().
+ Fixes #623663.
- * gst-libs/gst/interfaces/mixer.h:
- Add padding vars in place of the signal pointers
- when building with DISABLE_DEPRECATED so that the
- interface structure doesn't change size.
+2010-07-16 18:51:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-20 Jan Schmidt <thaytan@noraisin.net>
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gststreamsynchronizer.c:
+ playsink/streamsynchronizer: Remove and deactivate pads after calling the change_state function of the parent class
+ Fixes some deadlocks.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * ext/alsa/gstalsamixer.c:
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixerelement.c:
- * ext/alsa/gstalsamixertrack.c:
- * gst-libs/gst/interfaces/mixer.c:
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/interfaces/mixeroptions.c:
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/mixertrack.c:
- * gst-libs/gst/interfaces/mixertrack.h:
- * tests/check/Makefile.am:
- * tests/check/libs/mixer.c:
+2010-07-16 18:25:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
- Fixes: #152864
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Drop DISCONT flag on first buffer for new streams
+ Also reset stream state when going back to READY and on flush-stop.
- Add support for notifying mixer changes on the message bus, and
- implement it in alsamixer.
+2010-07-11 14:44:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- API: gst_mixer_get_mixer_flags
- API: gst_mixer_message_parse_mute_toggled
- API: gst_mixer_message_parse_record_toggled
- API: gst_mixer_message_parse_volume_changed
- API: gst_mixer_message_parse_option_changed
- API: GstMixerMessageType
- API: GstMixerFlags
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gststreamsynchronizer.c:
+ * gst/playback/gststreamsynchronizer.h:
+ * gst/playback/test7.c:
+ playsink: Fix gapless playback in many non-simple scenarios
+ Before gapless playback failed when switching between audio-only,
+ video-only and audio-video files, when choosing different clocks
+ and when the different streams had different durations.
+ This is now handled by a helper element, which keeps track of the
+ running times of all streams and synchronizes them.
+ Fixes bug #602437.
-2007-07-20 Michael Smith <msmith@fluendo.com>
+2010-07-11 14:43:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
- xcontext->im_format is only for testing XShm support (as the header
- file comments document). Use xvimage->im_format for everything else.
- Avoids spurious warnings on buffer allocation before setcaps.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Remove QOS event adjustments for gapless playback mode
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2010-07-09 17:15:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/volume/Makefile.am:
- * tests/icles/Makefile.am:
- We should use $(LIBM).
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: Post clock-provide and clock-lost messages when going from/to PLAYING
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2010-07-09 17:15:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/Makefile.am:
- This needs -lm.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: Post clock-provide and clock-lost messages when going from/to PLAYING
-2007-07-16 Wim Taymans <wim.taymans@gmail.com>
+2010-07-08 16:11:12 +0200 Philip Jägenstedt <philipj@opera.com>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_set_property),
- (gst_basertppayload_get_property):
- Don't break ABI, restore previous ranges. Keep the default random
- selection of timestamp and seqnum offset but as soon as the app sets a
- specific value, use that one.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: only associate .webm with WebM
+ .weba (audio) and .webv (video) were speculation on my part before
+ the public launch. As of yet no decision has been made on the
+ file extension for audio-only WebM, and I'm pretty sure there will
+ never be one for video-only.
+ Fixes bug #623837.
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2010-07-08 09:54:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Bastien Nocera <hadess at hadess dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: Use new gst_audio_clock_new_full()
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- Add option to turn off double-buffering for debugging purposes.
- Fixes #437169.
+2010-07-08 09:54:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: Use new gst_audio_clock_new_full()
- Patch by: Jorn Baayen <jorn at openedhand dot com>
+2010-07-08 08:32:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_set_property), (gst_ximagesink_get_property),
- (gst_ximagesink_init), (gst_ximagesink_class_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- add 'handle-expose' property. Useful for video widgets which may want to
- be in control of Expose behaviour. Fixes #380625
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * win32/common/libgstaudio.def:
+ audioclock: API: Add gst_audio_clock_new_full() with a GDestroyNotify for the user_data
+ Elements usually use their own instance as instance data but the
+ clock can have a longer lifetime than their elements and the clock
+ doesn't own a reference of the element.
+ Fixes bug #623807.
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2010-07-04 20:29:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_event), (gst_basertppayload_push),
- (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Fix ranges of rtp payloader properties so that the full range can be
- used in addition to -1 (random).
- Fix wrong seqnum reporting in caps.
- Fixes #420326.
+ * ext/theora/Makefile.am:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraenc.h:
+ theoraenc: Implement two pass encoding
+ Fixes bug #621349.
-2007-07-13 Wim Taymans <wim.taymans@gmail.com>
+2010-07-04 20:14:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_video_rate_init),
- (gst_video_rate_query):
- Use boilerplate.
- Add latency query, might not be perfect yet but already works a lot
- better. Fixes #442557.
+ * configure.ac:
+ * ext/theora/gsttheoraenc.c:
+ configure: Require libtheora >= 1.1
+ It's more than a year old at the time of the next -base release,
+ has many encoder and decoder improvements and gets us rid of a lot
+ of #ifdefs
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-04 20:08:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_setcaps):
- * sys/xvimage/xvimagesink.h:
- After a caps change, redraw our borders to avoid garbage left there
- when the image format changes to a smaller size, like 16:9 -> 4:3
- Also, hold the flow_lock a bit longer in the set_caps while we're
- fiddling with the xcontext.
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ theora: Use PROP_ instead of ARG_ for property enum values
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2010-05-04 12:09:57 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * Makefile.am:
- * configure.ac:
- * tests/Makefile.am:
- Remove bogus check for libcheck, since we check for
- gstreamer-check and it pulls in the required info from there, and we
- weren't actually _using_ the information for libcheck ourselves
- anyway.
+ * gst/playback/gstplaysink.c:
+ playsink: use proper error message code for failing state change
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-16 11:24:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- Fix the r_mask test for RGBA32 on little-endian.
- Fix a stupid typo that would have obviously broken
- compilation on big-endian, if anyone was testing.
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development
-2007-07-12 Wim Taymans <wim.taymans@gmail.com>
+=== release 0.10.30 ===
- * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
- (paint_hline_str4):
- * gst/videotestsrc/videotestsrc.h:
- Add alpha to the color struct.
- Use a default alpha value of 255 instead of 128.
+2010-07-15 01:20:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-12 Wim Taymans <wim.taymans@gmail.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.30
- * gst/playback/gstplaybasebin.c: (no_more_pads_full),
- (setup_source):
- Clear the dynamic pads counter when starting a new uri. This makes
- reusing playbin work again.
- Fixes #454264.
+2010-07-15 00:32:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-12 Stefan Kost <ensonic@users.sf.net>
+ * po/cs.po:
+ * po/lv.po:
+ po: update translations
- * configure.ac:
- Use pkg-config to locate check.
+2010-07-14 12:59:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Disconnect and destroy uridecodebins when going from READY to NULL
+ Fixes spurious errors that happen after an error and playing a working
+ stream afterwards or signals that are emitted for non-active groups.
+ Fixes bug #624266.
- * configure.ac:
- * tests/check/elements/volume.c: (GST_START_TEST):
- Fix 'make check' build against core CVS.
+2010-07-08 14:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-10 Stefan Kost <ensonic@users.sf.net>
+ * docs/design/Makefile.am:
+ docs: dist more of the gst-plugin-base design docs
- * gst-libs/gst/interfaces/propertyprobe.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Make gtk-doc happy.
+2010-07-07 00:35:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-08 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.29.4 pre-release
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_callback):
- Quick hack to make audiosinks stop at EOS when operating in
- pull-mode; needs to be fixed properly some day.
+2010-07-07 00:24:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-06 Stefan Kost <ensonic@users.sf.net>
+ * po/LINGUAS:
+ * po/es.po:
+ * po/fr.po:
+ * po/it.po:
+ * po/nl.po:
+ * po/pt_BR.po:
+ * po/sl.po:
+ * po/sv.po:
+ po: update translations
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix location of includes in the docs.
+2010-07-06 09:47:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstplaybin2.c:
+ Revert "playbin2: Don't put "raw" subtitle types in the raw caps for decodebin2"
+ This reverts commit 9d7538247ff1bf9841b53eeb71ddc47f2c662415.
+ If the DVD subpicture caps are not part of the raw caps, uridecodebin
+ doesn't qualify resindvdbin as raw source and plugs decodebins, which
+ causes broken DVD playback because of bugs elsewhere.
+ This change was originally added to only expose supported, raw subtitles,
+ e.g. if the subtitle sink did not support DVD subpictures but a converter
+ to some supported format exists. It's not very important right now because
+ we have nothing (that is autoplugged) to convert from plaintext/pango-markup
+ or DVD subpictures to something else.
+ Fixes bug #623583.
+
+2010-07-04 17:27:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
- (img_get_alpha_info):
- Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
- of the existing BGRA32 and RGBA32 formats with the alpha at the other
- end of the word. Partially fixes #451908
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Use correct Y offset for YVYU -> RGB conversions
+ Fixes bug #623530.
-2007-07-05 Stefan Kost <ensonic@users.sf.net>
+2010-07-04 17:26:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Simplify --extra-dir as gtkdoc scans recursively.
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Use correct Y offset for the YUY2 -> RGB conversions
+ Fixes bug #623530.
-2007-07-03 Wim Taymans <wim.taymans@gmail.com>
+2010-07-04 14:55:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
- (gst_adder_request_new_pad):
- Make getcaps more robust by not using the proxycaps function. This makes
- sure that we don't end up recursively calling getcaps upstream.
- See #316248.
+ * docs/plugins/inspect/plugin-ogg.xml:
+ docs: update ogg introspection info after riff fourcc addition
-2007-06-29 Wim Taymans <wim.taymans@gmail.com>
+2010-07-02 20:09:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/audioconvert.c:
- Include math.h to fix compilation.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix conversion of packed 4:2:2 YUV to 8 bit grayscale
+ The last pixel wasn't written before for odd widths.
+ Fixes bug #623418.
-2007-06-29 Jan Schmidt <thaytan@mad.scientist.com>
+2010-07-02 14:56:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
- Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
- format, as produced by some dc1394 cameras like the iSight.
- See http://www.fourcc.org/yuv.php#IYU1
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix conversion of packed 4:2:2 YUV to RGB
+ The last pixel wasn't written before.
+ Fixes bug #623384.
-2007-06-28 Sebastian Dröge <slomo@circular-chaos.org>
+2010-07-02 13:59:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
- (check_default), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_convert):
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_dithering_get_type),
- (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
- (gst_audio_convert_init), (gst_audio_convert_set_caps),
- (gst_audio_convert_set_property), (gst_audio_convert_get_property):
- * gst/audioconvert/gstaudioconvert.h:
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_setup_noise_shaping),
- (gst_audio_quantize_free_noise_shaping),
- (gst_audio_quantize_setup_dither),
- (gst_audio_quantize_free_dither),
- (gst_audio_quantize_setup_quantize_func),
- (gst_audio_quantize_setup), (gst_audio_quantize_free):
- * gst/audioconvert/gstaudioquantize.h:
- Implement dithering and noise shaping in audioconvert. By default now
- TPDF dithering (and no noise shaping) will be used when converting
- from a higher bit depth to 20 bit depth or smaller, otherwise
- everything will be as it is now.
- For the last audioconvert in a pipeline it would make sense to
- use some kind of noise shaping, enabling it by default for all
- conversions would give undesired results though. Fixes #360246.
- * tests/check/elements/audioconvert.c: (setup_audioconvert),
- (GST_START_TEST):
- Adjust unit test for the new audioconvert.
-
-2007-06-28 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
- Use other metrics as well when estimating the buffer level.
-
-2007-06-28 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
- Small debug improvement.
-
- * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
- (plugin_init):
- Tweak the rate estimation period.
- When calculating the buffer filledness in rate estimation mode, don't
- mix it with other metrics.
-
-2007-06-28 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
- (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
- When creating the groups, allow for a 5 second, unlimited buffers
- preroll phase after which we expose the group.
- When the group is exposed, use a small number of buffers up to a 2
- second limit. Also disconnect the overrun signal from multiqueue when we
- exposed the group because it is not needed anymore.
-
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
- Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
- to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
- (#451707); also, output some debugging info when dealing with
- freeform strings.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix invalid memory accesses with odd widths/heights during subsampling
+ Fixes bug #623375.
- * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
- Add unit test for the above.
+2010-07-01 21:21:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: If setup of the source element fails in READY->PAUSED deactive the current group
+ Otherwise the uridecodebin will be still a child of playbin2 and
+ its signals will still be connected. In future state changes this
+ will then emit unrelated signals that will confuse playbin2 or,
+ even worse, cause crashes and assertions.
+ Fixes bug #623318.
- * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
- Add description for Windows Media RTP caps.
+2010-06-30 21:20:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- Remove RTP fields that don't define the format from caps.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add FLV4 fourcc and map it to video/x-vp6-flash
+ Fixes #623176.
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-30 15:13:10 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
- Skip empty buffers, but not empty header buffers. That way the original
- vorbisdec unit test still passes (#451145); also, take into account
- that those empty packets might carry a granulepos.
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: declare with G_DEFINE_TYPE for type safety
+ Fixes #623233.
- * tests/check/Makefile.am:
- * tests/check/elements/vorbisdec.c:
- (_create_codebook_header_buffer), (_create_audio_buffer),
- (GST_START_TEST), (vorbisdec_suite):
- Add unit test that sends an empty packet.
+2010-06-24 16:55:57 +0200 Fredrik Söderquist <fs@opera.com>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle errors from _get_next_page in _do_seek.
+ If the source element failed here, oggdemux would crash.
+ Fixes #623218.
-2007-06-27 Wim Taymans <wim@fluendo.com>
+2010-06-30 11:00:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
- Don't error out on 0-sized packets, just emit a warning because this is
- not a fatal error. Fixes #451145.
+ * configure.ac:
+ configure: keep things sorted alphabetically
+ On special request. Because it's important, apparently.
-2007-06-25 Stefan Kost <ensonic@users.sf.net>
+2010-06-29 18:48:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.signals:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-decodebin2.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
- Update docs with caps info.
-
-2007-06-25 Tim-Philipp Müller <tim at centricular dot net>
-
- * po/POTFILES.in:
- Add more files with translatable strings (#450875).
-
-2007-06-23 Edward Hervey <edward@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
- The chain should be freed if we error out here, else it will leak.
- * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
- (cleanup_decodebin):
- Don't forget to *properly* remove the signals, else it will leak.
-
-2007-06-22 Jan Schmidt <thaytan@noraisin.net>
-
- * MAINTAINERS:
- Updating all the maintainers files
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.29.3 pre-release
-2007-06-21 Stefan Kost <ensonic@users.sf.net>
+2010-06-29 18:46:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
- (main):
- Destroy and recreate parse-launch based pipeline after stop to be able
- to play again. Reorder some code and add more comments.
+ * configure.ac:
+ configure: fix --disable-external
-2007-06-20 Jan Schmidt <thaytan@mad.scientist.com>
+2010-06-28 15:43:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- When handling a delayed-caps notification case, mark
- the group as dynamic so that the nbdynamic count is
- incremented and decremented correctly. Fixes: #449156
- Patch by: Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ * configure.ac:
+ Bump automake requirement to 1.10
+ For maintainability reasons and $(builddir).
+ Fixes #622944.
-2007-06-19 Andy Wingo <wingo@pobox.com>
+2010-06-27 10:43:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_init): Enable pull-mode operation.
+ * tests/examples/seek/jsseek.c:
+ * tests/examples/seek/seek.c:
+ examples: Remove some #if GTK_CHECK_VERSION(2,12,0)
+ We depend on GTK+ >= 2.14 already.
-2007-06-19 Michael Smith <msmith@fluendo.com>
+2010-06-26 21:28:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Change minimum rate back to 1000 to allow low-sample-rate wav files
- to play back.
+ * gst/videotestsrc/Makefile.am:
+ videotestsrc: Explicitely link with $(LIBM)
-2007-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-06-26 21:27:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/vi.po:
- Update translations.
+ * gst/videoscale/Makefile.am:
+ videoscale: Explicitely link with $(LIBM)
-2007-06-15 David Schleef <ds@schleef.org>
+2010-06-26 18:19:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstqueue2.c:
- Fix compile error from ignored return value.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.29.2 pre-release
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2010-06-26 18:19:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
- Update tmpbuf for all neccesary rows, not just one, as is required
- when downscaling.
- Fixes #402076.
+ * gst/adder/gstadderorc-dist.c:
+ * gst/adder/gstadderorc-dist.h:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.h:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ gst: update orc files
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2010-06-26 18:19:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
- (eos_buffer_probe):
- Add a test that ensures we set DELTA_UNIT on all non-header,
- non-video buffers, if we have a video stream.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_process_best_pad):
- Move setting delta_pad to earlier, where we inspect all pads, so
- that leading audio pages don't get DELTA_UNIT unset if they come
- before the first DELTA_UNIT from video pages. Fixes the newly-added
- test. Fixes #385527.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
-2007-06-14 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-26 17:55:12 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/pipelines/streamheader.c: (streamheader_suite):
- Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
- fails on the p5-ppc64 build bot and the failure looks like it is due
- to the same issue as #348114, ie. a compiler bug.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Properly clean DecodeChain after errors.
+ If an error happens, the PAUSED state will never be reached. If an
+ application re-uses decodebin2 (like totem) where one would normally
+ set to READY between each file, the cleanup that normally happens in
+ the PAUSED=>READY codepath will never be called, resulting in the
+ following file to re-use the previous demuxer/decoder/...
+ https://bugzilla.gnome.org/show_bug.cgi?id=622807
-2007-06-13 Edward Hervey <edward@fluendo.com>
+2010-06-26 12:39:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Fix build on MacOSX.
+ * docs/design/design-orc-integration.txt:
+ docs: fix a few typos
-2007-06-13 Wim Taymans <wim@fluendo.com>
+2010-06-26 12:03:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
- Fix compilation on mingw. Fixes #446972.
+ * tests/check/elements/videoscale.c:
+ checks: simplify GstBus usage in videoscale unit test
+ There's no need to run a main loop, add a bus watch and deal with
+ helper structs here just to wait for an EOS message.
-2007-06-12 Wim Taymans <wim@fluendo.com>
+2010-06-26 11:38:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * tests/check/elements/videoscale.c:
+ checks: speed up videoscale unit test a little
+ Use new gst_element_link_pads_full() function to link elements,
+ and disable all checks when linking (don't try this at home).
+ Down to 18s from 3m20s. Scary.
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_locked_enqueue):
- Fix a division by zero when the max percent is <= 0. Fixes #446572.
- also update the buffering status when receiving events. Fixes #446551.
+2010-06-25 17:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-06-11 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: check for ringbuffer state first
+ Check for the state of the ringbuffer before doing the checks of the other
+ buffer properties, when we're not started, we don't care about those values.
- Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+2010-06-24 13:30:59 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstqueue2.c: (gst_queue_peer_query),
- (gst_queue_handle_src_query):
- Wait for preroll before attempting to forward a duration query upstream.
- Fixes #445505.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use a more concise pad template
+ Speeds up caps nego 2 fold
+ https://bugzilla.gnome.org/show_bug.cgi?id=622696
-2007-06-07 Sebastien Moutte <sebastien@moutte.net>
+2010-06-24 15:31:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- Use G_GINT64_CONSTANT macro for int64 constant.
- * win32/common/libgstinterfaces.def:
- * win32/common/libgsttag.def:
- Add new exported functions.
+ * tests/icles/audio-trickplay.c:
+ tests: make audio-trickplay test compile when the gst debugging system is disabled
+ Fixes unused variable warning in that case.
-2007-06-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-24 15:13:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
- The BOS page of the first Dirac video stream needs to come before
- the BOS page of any Vorbis streams or other audio streams, just like
- it is with Theora.
+ * tests/check/gst/typefindfunctions.c:
+ tests: add test that runs all typefinders over random data
-2007-06-07 Wim Taymans <wim@fluendo.com>
+2010-06-06 12:31:35 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_get_range):
- Fix compilation.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: Mark ISO 14496-14 files as video/quicktime
+ These are currently being marked as audio/x-m4a which is incorrect.
+ https://bugzilla.gnome.org/show_bug.cgi?id=620720
-2007-06-06 Wim Taymans <wim@fluendo.com>
+2010-06-24 13:42:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * .gitignore:
+ .gitignore: add temporary orc test directory
- * gst/playback/gstqueue2.c: (gst_queue_init),
- (gst_queue_handle_sink_event), (gst_queue_chain),
- (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
- (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
- (gst_queue_src_activate_pull):
- Add pull based scheduling and fix some deadlocks. Fixes #444523.
- Does not yet completely work because duration queries upstream won't
- block yet.
+2010-06-24 13:30:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-06-06 Wim Taymans <wim@fluendo.com>
+ * tests/check/Makefile.am:
+ tests: add plugin loading whitelist to test environment
+ Only want to load core/-base plugins here.
- * configure.ac:
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Some more fseeko checks.
+2010-06-24 15:09:04 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-06-06 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 73ff93a to a519571
- * configure.ac:
- check for large file support.
+2010-06-24 08:41:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-06-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ tag: Fix printf format string
+ Use %s for strings, not %d.
- Based on a patch by Sven Arvidsson <sa at whiz dot se>:
+2010-06-24 12:06:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (parse_subrip),
- (subviewer_unescape_newlines), (parse_subviewer),
- (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
- * gst/subparse/gstsubparse.h:
- Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add a unit test for both SubViewer formats.
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix resampling of ARGB scanlines
+ Previously we would read behind the end of the source lines.
-2007-06-01 Michael Smith <msmith@fluendo.com>
+2010-06-16 14:08:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
- Don't overflow intermediate values when seeking to large time values
- in audiotestsrc.
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds GST_TAG_IMAGE_ORIENTATION mapping
+ Adds GST_TAG_IMAGE_ORIENTATION mapping to xmp helper lib.
+ Tests included.
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2010-06-16 11:19:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_have_data),
- (gst_queue_create_read), (gst_queue_read_item_from_file),
- (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
- Include stdio to define fseeko.
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * tests/check/libs/tag.c:
+ * win32/common/libgsttag.def:
+ tag: exif: Adds mapping for GST_TAG_IMAGE_ORIENTATION
+ Adds GST_TAG_IMAGE_ORIENTATION to the exif helper lib mapped tags.
+ Tests included.
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2010-06-23 12:10:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Edward Hervey <edward@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playsink: clear ts-offset pointer
+ We need to clear the pointer to our ts-offset element when we destroy the video
+ chain elements to make sure nobody derefs it to invalid memory afterwards.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
- (gst_v4lsrc_query):
- Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
+2010-06-23 10:16:07 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-06-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Reset ts_offset field when freeing chain
+ Otherwise we would end up with a bogus ->audiochain->ts_offset field
+ which would cause segfaults/assertions when trying to modify the
+ 'ts-offset' property in update_av_offset().
+ Was easy to trigger when using a list of audio+video files mixed with
+ video-only files in totem.
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
- Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
- our own implementation.
+2010-06-18 16:37:14 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-06-05 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/adder.c:
+ * tests/check/elements/appsink.c:
+ * tests/check/elements/audiotestsrc.c:
+ * tests/check/elements/gdpdepay.c:
+ * tests/check/elements/gdppay.c:
+ * tests/check/elements/multifdsink.c:
+ * tests/check/elements/videotestsrc.c:
+ * tests/check/elements/vorbisdec.c:
+ tests: use our own macros for the tests main function
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_change_state):
- Handle timestamp wraparound.
-
-2007-06-05 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gsturidecodebin.c: (no_more_pads_full),
- (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
- (gst_uri_decode_bin_change_state):
- Make sure we name srcpads uniquely even when using different internal
- decodebins.
- Signal no-more-pads when no more dynamic elements exist.
- Remove pads on cleanup.
-
-2007-06-05 Wim Taymans <wim@fluendo.com>
-
- Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
-
- * gst/playback/gstqueue2.c: (gst_queue_class_init),
- (gst_queue_init), (gst_queue_finalize),
- (gst_queue_write_buffer_to_file), (gst_queue_have_data),
- (gst_queue_create_read), (gst_queue_read_item_from_file),
- (gst_queue_open_temp_location_file),
- (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_is_empty), (gst_queue_is_filled),
- (gst_queue_change_state), (gst_queue_set_temp_location),
- (gst_queue_set_property):
- Add support for filebased buffering. Fixes #441264.
-
-2007-06-05 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
- (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
- (caps_notify_group_cb), (gst_decode_group_new),
- (gst_decode_group_free):
- Add support for delayed caps fixation when autoplugging.
- Optimize cases where a multiqueue is not needed/wanted, like right after
- anything that is not a demuxer.
-
-2007-06-05 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
- (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
- consideratly speedup ogg chain detection by not trying to find a base
- timestamp for skeleton streams.
-
-2007-06-05 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
- (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
- (gst_multi_fd_sink_remove_flush),
- (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Add support for remove_flush.
+2010-06-18 14:17:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-06-05 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ tag: Use gst_tag_list_peek_string_index in vorbistag
+ Use _peek_string_index instead of _get_string_index to avoid
+ a string copy
- * docs/design/draft-keyframe-force.txt:
- * ext/theora/theoraenc.c: (theora_enc_sink_event),
- (theora_enc_chain):
- Add draft design for forcing keyframes in encoders and implement in
- theoraenc.
+2010-06-14 12:27:02 +0200 Philippe Normand <pnormand@igalia.com>
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * sys/ximage/ximagesink.c:
+ * sys/ximage/ximagesink.h:
+ ximagesink: Ask pad peer to accept new caps once only
+ In buffer_alloc, if the buffer caps are new, call
+ gst_pad_peer_accept_caps once only, it's useless to call it in the
+ cases where we know it will always fail.
+ Fixes bug #621190
- * configure.ac:
- Back to CVS
+2010-06-17 17:07:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.13 ===
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add YUY2/YVYU to all RGB formats conversions
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+2010-06-17 16:57:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- releasing 0.10.13, "What's Going on?"
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y42B to YUY2/YVYU/UYVY conversion for odd widths
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
+2010-06-17 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- In riff, the depth is stored in the size field but it just means that
- the least significant bits are cleared. We can therefore just play
- the sample as if it had a depth == width. Fixes: #440997
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix YUY2/YVYU/UYVY to Y42B conversion for odd widths
- Patch by: Wim Taymans <wim@fluendo.com>
- Patch by: Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-17 16:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
+ * common:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ docs: update introspected plugin docs for gstdoc-scangobj and other changes
+ Update common for latest gstdoc-scangobj and inspect xml files for
+ escaping and pad template order changes. Update other gtk-doc files
+ for API additions and object hierarchy changes.
- * gst-libs/gst/floatcast/floatcast.h:
- Define inline when needed on win32 builds. Fixes: #441295
- Patch by: Sebastien Moutte <sebastien@moutte.net>
+2010-06-16 19:15:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-29 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: improve autoplugging
+ Use the pad caps when they are available to continue the autoplugging. If the
+ pad caps are set, they are fixed and then we can directly continue autoplugging.
- * gst/playback/gstplaybasebin.c: (queue_overrun),
- (no_more_pads_full):
- Stop buffering when the group is commited because the queues filled up.
- Fixes #442024.
+2010-06-15 16:49:17 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-25 Jan Schmidt <thaytan@mad.scientist.com>
+ * common:
+ Automatic update of common submodule
+ From 9339ccc to 35617c2
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
- (gst_alsa_mixer_free), (gst_alsa_mixer_update),
- (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
- (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
- (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_interface_supported),
- (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
- (gst_alsa_mixer_element_set_property),
- (gst_alsa_mixer_element_get_property),
- (gst_alsa_mixer_element_change_state):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
- * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
- (gst_mixer_option_changed):
- * gst-libs/gst/interfaces/mixer.h:
- Revert commits towards #152864 made so far. We'll pick it up again
- after the 0.10.13 release.
+2010-06-15 16:53:49 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-05-24 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 5adb1ca to 9339ccc
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- After an interrupt (PAUSED/flush) assume that the next sample should not
- be aligned to the previous sample. Fixes #417992.
+2010-06-15 16:34:54 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-05-24 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 57c89b7 to 5adb1ca
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Don't add channels and rate fields to the template caps for
- audio/x-dts, as wavparse might not always be able to set them,
- which would then lead to 'caps are not a real subset of the
- template caps' warnings.
+2010-06-15 15:32:34 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+ * common:
+ Automatic update of common submodule
+ From c804988 to 57c89b7
- * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
- Handle unknown or invalid pads without crashing, as might occur if
- a media file like an mp3 is specified as a subtitle file.
- Fixes: #410039
+2010-06-15 13:09:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/check/elements/audioresample.c:
+ Revert "audioresample: set pads as negotiable"
+ This reverts commit 5f74f3a82eb54f9a9517f99dffbe45ce4d474870.
- * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
- (setup_sinks):
- Block the subtitle bin output queue before ghosting it and linking,
- then unblock after. This avoids spurious not-linked errors caused
- by the queue starting up (because it gets linked when it is ghosted).
- Fixes: #350299
+2010-06-15 13:09:29 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-23 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/check/elements/audioconvert.c:
+ Revert "audioconvert: set pads negotiable"
+ This reverts commit bbd7dee8f604bd0373a82e6e5cc3eec8313806ac.
- * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
- Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
- file. Avoids flukes where the input gets typefound to some valid but
- useless type.
+2010-06-14 15:19:32 -0700 David Schleef <ds@schleef.org>
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix black horizontal line in image
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
- (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
- Add unit test for gnomevfssink seeking and position reporting for
- file:// URIs.
+2010-06-14 15:05:16 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Init char variable for gps coordinates
+ Initialize char variable for gps coordinates deserialization to 0
+ to identify when it couldn't be parsed/found and error out.
+ Fixes #621509
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+2010-06-14 18:10:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
- (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
- (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
- * ext/gnomevfs/gstgnomevfssink.h:
- Fix position reporting, especially after a seek (from upstream),
- see #412648.
+ * tests/check/elements/audioconvert.c:
+ audioconvert: set pads negotiable
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-14 17:48:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/cdparanoia/gstcdparanoiasrc.c:
- Repair umlaut.
+ * tests/check/elements/audioresample.c:
+ audioresample: set pads as negotiable
-2007-05-22 Jan Schmidt <thaytan@mad.scientist.com>
+2010-06-14 16:25:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Specify the full valid range for MP3 samplerates. Fixes a regression
- caused by extra header checks since the last release.
+ * common:
+ Revert accidental downgrade of common revision.
-2007-05-21 Jan Schmidt <thaytan@mad.scientist.com>
+2010-06-14 16:07:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Fix a locking-order bug I introduced with my changes the other day.
- Patch by Mike Smith.
+ * tests/check/elements/videoscale.c:
+ videoscale: And only expect a single buffer in the unit test
-2007-05-21 Michael Smith <msmith@fluendo.com>
+2010-06-14 16:02:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_handle_data_packet):
- Don't look inside 0-length packets (which indicate duplicated
- frames)
+ * tests/check/elements/videoscale.c:
+ videoscale: Only convert one buffer instead of five
+ Should make the unit test a lot faster.
-2007-05-21 Wim Taymans <wim@fluendo.com>
+2010-06-14 14:13:32 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_read_sector):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Small cleanups.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix unitialized variables
+ yay macosx compilers :(
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix typo.
+2010-06-14 14:13:16 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- Add some FIXME
+ * gst-libs/gst/video/video.c:
+ video: Fix unitialized variable.
+ yay macosx compilers :(
- * gst/playback/gstdecodebin.c: (queue_underrun_cb):
- And some debug info when a FIXME path is hit.
+2010-06-14 13:27:01 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-21 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use Quarks for structure name/field checking
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_class_init),
- (gst_base_rtp_audio_payload_init),
- (gst_base_rtp_audio_payload_finalize),
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer),
- (gst_base_rtp_payload_audio_handle_event):
- Some cleanups, remove minptime property as it is now in the parent
- class.
- Override parent class event function.
+2010-06-14 13:26:02 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_event), (gst_basertppayload_set_property),
- (gst_basertppayload_get_property):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Add min-ptime property.
- Add handle-event vmethod. Fixes #415001.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Speed up _remove_format_info
+ Instead of copying full caps, use the fact that the provided caps only have
+ one structure and only copy around structures.
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+2010-06-14 13:24:06 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_change_state):
- Fix typo in comment.
+ * common:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Transfer structures instead of copying them
+ Avoids many expensive structure copies
- * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
- free_dynamics, pad_probe, close_pad_link, try_to_link_1,
- get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
- close_link):
- * gst/playback/gstplaybin.c (gst_play_bin_set_property,
- gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
- Remove trailing whitespaces in comments.
+2010-06-14 13:20:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/volume/Makefile.am:
- Fix tabs.
+ * configure.ac:
+ configure: Use GLIB_EXTRA_CFLAGS
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+2010-06-14 13:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ * common:
+ Automatic update of common submodule
+ From 7a0fdf5 to c804988
- * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
- set_option, get_option, _gst_reserved):
- Revert reordering functions (keep ABI).
+2010-06-14 11:31:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-17 Jan Schmidt <thaytan@mad.scientist.com>
+ * common:
+ Automatic update of common submodule
+ From 6da3bab to 7a0fdf5
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
- (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
- (gst_ximagesink_show_frame):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
- (gst_xvimagesink_show_frame):
- When we create our own window, indicate that we handle the
- WM_DELETE client message from the window manager, so that it won't
- kill our window (and our app) along with it. Handle ClientMessage,
- post an error on the bus, and close the window. Further buffers
- arriving will result in a FlowError because the window has been
- destroyed.
+2010-06-14 11:20:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Fixes: #393975
+ * tests/examples/seek/jsseek.c:
+ jsseek: Don't use deprecated GLib API
+ Fixes once again bug #605100.
- Clean up the X event handling loop and make them the same for
- both xvimagesink and ximagesink while I'm at it.
+2010-06-14 11:16:45 +0200 Prahal <prahal at yahoo.com>
-2007-05-17 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: use accumulator for autoplug-sort
+ Use an accumulator for the autoplug-sort signal so that we can stop the emission
+ when a signal handler produced a valid result. This avoids the object handler
+ to overwrite the results from user signals.
+ Fixes #621161
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
- Make decodebin2 autoplug depayloaders too.
+2010-06-14 11:11:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gsturidecodebin.c: (source_new_pad):
- Set the newly created decoder in a usable state when autoplugging a
- dynamic source such as RTSP.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: activate_chain must not be called with a NULL chain
+ It will crash later and shouldn't really happen anyway unless
+ something is really wrong.
-2007-05-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-14 11:08:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gststreaminfo.c: (cb_probe):
- Ignore video-codec tag for audio streams and ignore audio-codec tags
- for video streams. Should make codec name collection a bit more
- robust against sloppy demuxers that send tag events containing both
- tags down each pad.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfssrc: Fix possible NULL pointer dereference
+ It's always an error if gst_buffer_try_new_and_alloc() returns NULL
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2010-06-14 11:03:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (update_rates):
- Tweak the buffering thresholds a little.
- Update the buffer size with the previously calculate rate instead of
- only when we calculate a new rate so that we get smoother buffering
- updates.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Return FALSE from the seek handler if no seek callback was set
- * gst/playback/Makefile.am:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
- (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
- (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
- (gst_uri_decode_bin_get_property), (unknown_type),
- (add_element_stream), (no_more_pads_full), (no_more_pads),
- (source_no_more_pads), (new_decoded_pad), (array_has_value),
- (gen_source_element), (has_all_raw_caps), (analyse_source),
- (remove_decoders), (make_decoder), (remove_source),
- (source_new_pad), (setup_source), (decoder_query_init),
- (decoder_query_duration_fold), (decoder_query_duration_done),
- (decoder_query_position_fold), (decoder_query_position_done),
- (decoder_query_latency_fold), (decoder_query_latency_done),
- (decoder_query_seeking_fold), (decoder_query_seeking_done),
- (decoder_query_generic_fold), (gst_uri_decode_bin_query),
- (gst_uri_decode_bin_change_state), (plugin_init):
- New element that intergrates a source, optional buffering element and
- decodebin.
-
-2007-05-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-14 09:53:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Bump libtheora requirement to 1.0alpha5 for the pixformat check
- (also has a .pc file, so we don't need the fallback check any
- longer). Fixes #438840.
+ * ext/gio/gstgiostreamsrc.c:
+ giostreamsrc: Fix copy&paste error in the docs
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2010-06-14 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_get_type),
- (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
- (apply_segment), (apply_buffer), (update_buffering),
- (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_handle_sink_event), (gst_queue_is_filled),
- (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
- (plugin_init):
- fix build.
+ * ext/ogg/gstoggstream.c:
+ ogg: fix debug message printf format some more
+ Just cast the pointer diff, so it works everywhere without
+ warnings. Can't use %tu, because that modifier is C99. Warning
+ was: "format '%li' expects type 'long int', but argument 8 has
+ type 'int'".
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2010-06-13 22:17:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstqueue2.c: (gst_queue_get_type),
- (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
- (gst_queue_getcaps), (gst_queue_bufferalloc),
- (gst_queue_acceptcaps), (update_time_level), (apply_segment),
- (apply_buffer), (update_buffering), (reset_rate_timer),
- (update_rates), (gst_queue_locked_flush),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_handle_sink_event), (gst_queue_is_empty),
- (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
- (gst_queue_loop), (gst_queue_handle_src_event),
- (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
- (gst_queue_src_activate_push), (gst_queue_change_state),
- (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
- On our way to playbin2 this is the new network queue that does buffering
- all by itself using high and low watermarks. It can also measure up and
- downstream bandwidth to optimally size the queue.
-
-2007-05-17 Michael Smith <msmith@fluendo.com>
-
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
- Use the segment->last_stop value to calculate the next timestamp to
- generate after a seek; not the segment->start value.
+ * tests/check/elements/videoscale.c:
+ videoscale: Add ffmpegcolorspace after videotestsrc for the unit test
-2007-05-15 David Schleef <ds@schleef.org>
+2010-06-13 20:57:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/Makefile.am: Install docs even when --disable-gtk-doc
- is disabled. This matches the behavior of gtk+. Fixes #349099.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: ...and add Y16 case for the linear scaling
-2007-05-15 Wim Taymans <wim@fluendo.com>
+2010-06-13 20:38:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
- Some more chained streaming ogg timestamp fixes.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add Y16 case for 4-tap scaling
-2007-05-15 Wim Taymans <wim@fluendo.com>
+2010-06-13 18:27:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_handle_page):
- Add some FIXMEs.
- Fix chain start/stop segment handling based on patch by
- <ahalda at cs dot mcgill dot ca> see #320984.
+ * tests/check/Makefile.am:
+ tests: Fix linking of the tags test
-2007-05-15 Michael Smith <msmith@fluendo.com>
+2010-06-13 08:20:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- We don't require a C++ compiler. So don't require one.
+ * gst-libs/gst/video/video.h:
+ video: Re-add (but deprecated) GST_VIDEO_{RED,GREEN,BLUE}_MASK_1[56]
-2007-05-15 Stefan Kost <ensonic@users.sf.net>
+2010-06-12 21:04:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
- gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
- gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
- gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
- gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
- gst_alsa_mixer_update_track):
- Apply some of the cleanup Tim suggested in #152864 afterwards.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use correct variables for debug output
-2007-05-15 Stefan Kost <ensonic@users.sf.net>
+2010-06-12 16:51:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y16 from/to GRAY8 conversion
- * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
- _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
- gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
- gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
- gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
- gst_alsa_mixer_handle_source_callback,
- gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
- gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
- gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
- gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
- gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
- gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
- * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
- * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
- gst_alsa_mixer_element_interface_supported,
- gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
- gst_alsa_mixer_element_set_property,
- gst_alsa_mixer_element_get_property,
- gst_alsa_mixer_element_change_state):
- * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
- * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
- gst_mixer_option_changed):
- * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
- volume_changed, option_changed, _gst_reserved):
- Implement notification for alsamixer. Fixes #152864
+2010-06-12 16:31:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-14 David Schleef <ds@schleef.org>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Don't crash when doing gray YUV to GRAY conversion
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add support for video/x-raw-bayer.
+2010-06-12 16:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-12 David Schleef <ds@schleef.org>
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ videoscale: Update disted orc files
- * sys/xvimage/xvimagesink.c:
- Add some sanity checking for the XVImage size returned by X.
- Related to #377400.
+2010-06-12 16:16:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-12 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Allow video/webm for progressive downloading
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_set_gst_timestamp):
- Parse and use additional caps fields as described in updated
- application/x-rtp caps spec.
+2010-06-12 13:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-12 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for more gray formats
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_collect_chain_info):
- If there is a stream in a chain without any data packets, ignore the
- stream in the total length calculations. Might be related to #436820.
+2010-06-01 16:45:34 +0000 Martin Bisson <martin.bisson@gmail.com>
-2007-05-11 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video.{c,h}: Fix an endianness bug fix.
+ This commit makes sure the endianness is ok for RGB/BGR 15/16 formats.
- * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
- (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
- (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (plugin_init):
+2010-06-01 14:42:54 +0000 Martin Bisson <martin.bisson@gmail.com>
- Consolidate and re-work our mpeg system stream detection to probe
- more packets and produce a higher confidence result. Fixes a
- regression caused by lowering the typefind probability last year
- - related to bug #397810. Remove the redundant MPEG-1 specific
- typefind function, as the new one detects both MPEG-1 & MPEG-2
- happily.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video.{c,h}: Add support for RGB and BGR with 15 and 16 bits.
- Also cleanup the MPEG elementary and MPEG-TS detection functions a
- little.
+2010-06-12 13:35:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Tested against my media test directory, with some improvements and
- no regressions.
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Use libgstvideo for caps parsing, etc
-2007-05-10 Wim Taymans <wim@fluendo.com>
+2010-06-12 13:04:43 +0200 Philippe Normand <phil@base-art.net>
- * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
- (queue_out_of_data):
- Connect to the new queue "pushing" signal instead of the broken
- "running" one.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix format string compiler warning on OS X
-2007-05-09 Sebastien Moutte <sebastien@moutte.net>
+2010-06-12 13:00:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer):
- Move variable declaration before the first instruction.
- * gst/videotestsrc/videotestsrc.c:
- Define M_PI if it's not defined yet.
- * win32/common/libgstrtp.def:
- Add new exported functions.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use GST_VIDEO_CAPS_GRAY{8,16}
-2007-05-09 Michael Smith <msmith@fluendo.com>
+2010-06-12 12:57:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_handle_type_packet):
- gst_pad_push_event() does not return a GstFlowReturn!
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Implement linear merging of Y16 scanlines with orc
-2007-05-09 Wim Taymans <wim@fluendo.com>
+2010-06-12 08:26:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/scrubby.c: (stop_cb), (main):
- * tests/examples/seek/seek.c: (do_seek):
- Some small cosmetic changes.
+ * common:
+ Automatic update of common submodule
+ From 733fca9 to 6da3bab
-2007-05-08 Stefan Kost <ensonic@users.sf.net>
+2010-06-11 22:16:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
- gst_adder_change_state):
- * gst/adder/gstadder.h (bps, offset, collect_event, segment,
- segment_pending, segment_position, segment_rate):
- Handle playback-rate on adder.
+ * configure.ac:
+ * ext/cdparanoia/Makefile.am:
+ cdparanoia: check for cdparanoia with pkg-config first
+ cdparanoia now has a .pc file in post-0.10.2 SVN, so use
+ that to check for cdparanoia before we try all the other
+ checks. Besides being generally nicer, this may help with
+ correctly detecting cdparanoia on OSX some day (see #609918).
-2007-05-07 Michael Smith <msmith@fluendo.com>
+2010-06-11 12:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_reset),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_type_packet), (theora_dec_change_state):
- Don't push events (newsegment, tags) before initialising the
- decoder.
- This is neccesary for seeking to work correctly in gnonlin.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: look for dts frames at non-zero offsets too
+ Scan a bit into the data when checking for dts frames instead
+ of expecting the frame sync to be right at the start of the
+ data. This is needed for some dts-disguised-as-pcm-in-wav files.
+ See #413942.
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+2010-06-10 18:12:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst/adder/gstadder.c:
- * gst/audiotestsrc/gstaudiotestsrc.c
- (gst_audio_test_src_create_white_noise):
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
- VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
- volume_sink_template, volume_src_template, gst_volume_init,
- volume_process_double, volume_process_int16,
- volume_process_int16_clamp):
- Doc fixes and formatting.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add typefinder for dts audio
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-11 15:23:14 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/Makefile.am:
- * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
- Minimal check for volume's GstController usability; also another
- test for #422295.
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix unitialized variables
+ I hate thee macosx
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-11 08:47:27 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track):
- Fix it so that it (a) makes sense and (b) doesn't break
- everything cdda-related including the unit test.
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix debug statements
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+2010-06-11 08:47:17 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track):
- Fix build when disabling asserts.
+ * gst-libs/gst/tag/gstexiftag.c:
+ exiftag: Fix unitialized variable
-2007-05-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-10 20:45:42 +0300 Stefan Kost <ensonic@users.sf.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
- When XShm is not available, we might get row strides that are not
- rounded up to multiples of four; this is bad, because virtually
- every RGB-processing element in GStreamer assumes rowstrides are
- rounded up to multiples of four, so let's allocate at least enough
- memory to avoid crashes in this case. The image will still be
- displayed distorted though if this happens, so that still needs
- fixing (maybe by allocating a bigger image with an 'even' width
- and then clipping it appropriately when rendering - something for
- Xlib aficionados in any case).
+ * win32/common/libgsttag.def:
+ win32: update def file
-2007-05-03 Michael Smith <msmith@fluendo.com>
+2010-06-10 20:36:32 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- If a buffer doesn't have a timestamp, assume it's contiguous with
- the previous buffer, and synthesise timestamps appropriately.
+ * gst-libs/gst/tag/tag.h:
+ docs: fix gtk-doc warnings
+ Variable names in function prototypes in the headers should match the doc-
+ comment.
-2007-05-03 Edward Hervey <edward@fluendo.com>
+2010-06-10 08:47:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/videorate.c: (GST_START_TEST):
- Set buffer timestamp to a valid value in order to test the buffer
- really does stay in videorate.
+ * gst/playback/gstplaybin2.c:
+ playbin2: If the text-sink claims to support ANY caps assume it only support raw plaintext subtitles
+ Fixes bug #621071.
-2007-05-03 Edward Hervey <edward@fluendo.com>
+2010-06-10 08:46:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
- There is no sensible way to handle incoming buffers which don't have a
- valid timestamp. We therefore discard them and wait for the next one.
+ * tests/icles/playbin-text.c:
+ icles: Only accept plain subtitles in the playbin-text icles test
-2007-05-01 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-09 22:34:24 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
- * gst/playback/gstdecodebin2.c: (plugin_init):
- Better error message for text files.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for VP6F (On2 VP6 Flash variant)
-2007-04-29 Wim Taymans <wim@fluendo.com>
+2010-06-09 12:35:40 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
- Fix offset bug in generation RR packets.
+ * configure.ac:
+ Use the Orc m4 macro
-2007-04-27 Julien MOUTTE <julien@moutte.net>
+2010-06-09 12:40:00 -0700 David Schleef <ds@schleef.org>
- * ext/theora/theoradec.c: (_theora_granule_time),
- (theora_dec_push_forward), (theora_handle_data_packet),
- (theora_dec_decode_buffer): Calculate buffer duration correctly
- to generate a perfect stream (#433888).
- * gst/audioresample/gstaudioresample.c:
- (audioresample_check_discont): Glib provides ABS.
+ * common:
+ Automatic update of common submodule
+ From fad145b to 733fca9
-2007-04-27 Wim Taymans <wim@fluendo.com>
+2010-06-09 12:33:51 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
- (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
- (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
- (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
- (gst_rtcp_packet_bye_set_reason):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Fix RB block parsing and writing.
- Add support for constructing BYE packets.
+ * common:
+ Automatic update of common submodule
+ From 47683c1 to fad145b
-2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-09 15:58:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
- (gst_base_audio_src_create):
- * po/POTFILES.in:
- When posting a warning message because samples were dropped, post
- something more intelligible than he default error message for clock
- errors which is just confusing in this context (#432984).
+ * gst-libs/gst/tag/gstexiftag.c:
+ tag: exif: Refactor byte-order handling
+ Only check for valid byte-order values when creating the
+ exif readers and writers
-2007-04-25 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
- (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
- (read_packet_header), (gst_rtcp_packet_move_to_next),
- (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
- (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
- (gst_rtcp_packet_sdes_get_item_count),
- (gst_rtcp_packet_sdes_first_item),
- (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
- (gst_rtcp_packet_sdes_first_entry),
- (gst_rtcp_packet_sdes_next_entry),
- (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
- (gst_rtcp_packet_sdes_add_entry):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Implement code to write SR, RR and SDES packets.
+2010-05-10 14:01:46 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstexiftag.c:
+ * tests/check/libs/tag.c:
+ tag: exif: Adds new geo-location tag mappings
+ Adds mappings for:
+ GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION
+ GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
+ GST_TAG_GEO_LOCATION_MOVEMENT_SPEED
+ GST_TAG_GEO_LOCATION_ELEVATION
+ Does some refactoring in the code to reduce number of parameters
+ passed to functions
+ Tests included.
+
+2010-04-04 22:25:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
+ * tests/check/libs/tag.c:
+ tests: tag: Adds unit tests for exif helper lib
+ Adds some simple unit tests for exif helper lib functions
+ Fixes #614872
- * sys/ximage/ximagesink.c:
- Fix build if XShm is not available (#432362).
+2010-04-03 23:02:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-04-24 Sebastian Dröge <slomo@circular-chaos.org>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstexiftag.c:
+ * gst-libs/gst/tag/tag.h:
+ * win32/common/libgsttag.def:
+ tag: Adds basic exif tags support
+ Adds exif helper lib functions to parse exif buffers from/to
+ taglists. Exif is tipically used in jpeg images, but it can
+ also be embedded into TIFF, AVI and WAV formats.
+ Adds a couple function to handle exif in tiff header structures, that is how
+ exif is embedded in jpeg and (obviously) in tiff.
+ API: gst_tag_list_to_exif_buffer
+ API: gst_tag_list_to_exif_buffer_with_tiff_header
+ API: gst_tag_list_from_exif_buffer
+ API: gst_tag_list_from_exif_buffer_with_tiff_header
+ Fixes #614872
+
+2010-06-09 17:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
- Initalize the AudioConvertCtx with zeroes, otherwise it will contain
- pointers to random memory which are passed to g_free() when
- audio_convert_prepare_context() is called the first time.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle SEEKING query in push mode too
-2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-09 16:38:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Dan Williams <dcbw redhat com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Update the total time from the Skeleton 4 indexes
+ Fixes bug #620939, see bug #607945.
- * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
- Don't leak incoming buffer if gst_pad_push() returns a
- non-OK flow. Fixes #432755.
-
- * tests/check/elements/videorate.c: (GST_START_TEST),
- (videorate_suite):
- Unit test for the above by Yours Truly.
+2010-06-09 16:33:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-23 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggstream.c:
+ oggstream: Implement latest version of the Skeleton 4.0 spec
+ Fixes bug #620939.
- * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
- (gst_adder_sink_event), (gst_adder_collected):
- Fix non-flushing segmented seeks, Fixes #340060 for me
+2010-06-09 16:59:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/volume/gstvolume.c:
+ volume: make the orc codes available for testing.
+ Add a USE_ORC define for now and switch 'this' to 'self'. Having orc enabled
+ passes the test suite and various manual gst-launch pipelines.
- Patch by: Olivier Crete <tester at tester ca>
+2010-06-08 13:34:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_class_init),
- (gst_base_rtp_audio_payload_init),
- (gst_base_rtp_audio_payload_dispose):
- Chain up to parent class in dispose function; get rid of
- unnecessary 'diposed' flag in private structure (#415001).
+ * .gitignore:
+ .gitignore: add orc-related temp files
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-08 13:26:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs.types:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_class_init):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Some minor docs fixes and additions; also add missing 'Since' bits.
+ * configure.ac:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/gstaudioresample.c:
+ Fix build if orc is not installed
+ Orc is not a hard requirement. Things should still compile and
+ work without orc, but slow fallback code may be used in this
+ case. Fix up configure to not error out if orc is not installed
+ and wrap use of orc profiling in audioresample in #ifdefs.
+ Fixes #620136 some more.
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-08 13:11:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Zeeshan Ali <zeenix gmail com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Implement correct parsing of Skeleton 4.0 index packets
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer),
- (gst_base_rtp_audio_payload_push):
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- The recently-added gst_base_rtp_audio_payload_push() should take an
- object of type GstBaseRTPAudioPayload as first argument (#431672).
+2010-06-08 12:01:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Add parsing of Skeleton 4.0 indexes
- * gst/audioresample/gstaudioresample.c:
- Make more functions static, just because we can.
+2010-06-08 11:40:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Parse segment length and content offset from fishead
+ And print them for debugging purposes. Not sure if we can do anything useful
+ with this information.
- * tests/check/elements/audioresample.c:
- Add unit test for audioresample shutdown crasher (#420106).
+2010-06-08 11:31:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-20 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Parse Skeleton stream major/minor version
- * gst/subparse/gstsubparse.c:
- * gst/subparse/samiparse.c:
- Use GST_DISABLE_XML here
+2010-06-08 11:26:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_navigation_send_event):
- * sys/xvimage/xvimagesink.h:
- Include stdlib.h when using atoi.
-
- * tests/check/elements/playbin.c: (playbin_suite):
- Use GST_DISABLE_REGISTRY here
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Use binary search for searching in the index
-2007-04-19 Michael Smith <msmith@fluendo.com>
+2010-06-08 11:02:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
- (theora_enc_sink_event), (theora_enc_change_state):
- Track initialisation state; don't try to use encoder state if we're
- not initialised (it'll segfault).
+ * tests/check/libs/video.c:
+ video: Fix unit test, the Y800 checks were not used before and were not working
-2007-04-18 Stefan Kost <ensonic@users.sf.net>
+2010-06-08 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/.cvsignore:
- Fix build.
+ * gst-libs/gst/video/video.c:
+ video: Return TRUE in gst_video_format_is_gray() for Y800 and Y16
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-08 00:33:31 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Allow random depths between 1 and 32 instead of only multiplies of 8.
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ audioconvert, videotestsrc: Update generated Orc code
+ Fixes compile errors with initialization of unions.
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-08 00:32:36 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Set the maximum number of channels for PCM and float in the correct
- place to have it also used when creating the template caps.
+ * REQUIREMENTS:
+ requirements: change liboil to Orc
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-06 23:50:05 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Correctly support 4, 6 and 8 channels with normal PCM and float
- wav files.
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: convert from liboil to orc
- Fix the depth and signedness calculation in extensible wav files and
- also handle 1, 2, 4, 6, 8 channels here when a file without channel
- mask is found.
+2010-06-06 23:48:35 -0700 David Schleef <ds@schleef.org>
- Add support for float, alaw and mulaw in extensible wav files.
+ * tests/check/Makefile.am:
+ tests: Add orc tests
- This allows correct playback of all but 5 files from
- http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
-
- (gst_riff_create_audio_template_caps):
- Add voxware and float formats to the template caps.
+2010-06-06 23:48:15 -0700 David Schleef <ds@schleef.org>
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/volume/Makefile.am:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.h:
+ * gst/volume/gstvolumeorc.orc:
+ volume: convert from liboil to orc
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+2010-06-06 23:47:53 -0700 David Schleef <ds@schleef.org>
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
- Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
+ * gst/videotestsrc/Makefile.am:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.h:
+ * gst/videotestsrc/gstvideotestsrcorc.orc:
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: convert from liboil to orc
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
- Use the correct format strings for integer formats.
+2010-06-06 23:47:16 -0700 David Schleef <ds@schleef.org>
-2007-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ videoscale: convert from liboil to orc
- * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
- Don't use pad_alloc_buffer_and_set_caps to create a small header
- packet, or, worse, to create a big temporary video buffer using the
- src pad.
+2010-06-06 23:46:41 -0700 David Schleef <ds@schleef.org>
-2007-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/audioconvert/Makefile.am:
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/audioconvert/gstaudioconvertorc-dist.h:
+ * gst/audioconvert/gstaudioconvertorc.orc:
+ audioconvert: convert from liboil to orc
- * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
- * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
- GST_START_TEST, buffer_probe_cb, GST_START_TEST):
- Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
+2010-06-06 23:45:58 -0700 David Schleef <ds@schleef.org>
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/adder/Makefile.am:
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ * gst/adder/gstadderorc-dist.c:
+ * gst/adder/gstadderorc-dist.h:
+ * gst/adder/gstadderorc.orc:
+ adder: convert from liboil to orc
- * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
- GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
- streamheader_suite):
- Add another test set up for failure
+2010-06-06 23:45:10 -0700 David Schleef <ds@schleef.org>
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/design/Makefile.am:
+ * docs/design/design-orc-integration.txt:
+ docs: Add notes about Orc integration
- * tests/check/Makefile.am:
- * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
- GST_START_TEST, streamheader_suite, main):
- Add a test for the streamheader bug Wim fixed.
+2010-06-06 23:34:39 -0700 David Schleef <ds@schleef.org>
-2007-04-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ configure: convert liboil check to orc
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix misleading comment.
+2010-06-08 07:34:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-13 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggmux.c:
+ oggmux: Start a new page for every CMML buffer
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- More sanity checks for the header fields.
+2010-06-07 14:38:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: add av-offset property
+ Add av-offset property to control the audio and video sync offset. This can be
+ used to to manually correct badly synced streams.
+ See #620529
- * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
- Try encodings from all environment variables, not just those in the
- first environment variable that is set.
+2010-06-07 08:31:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-12 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: Map "Y8 " and "GREY" to "Y800" and add it to the template caps
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
- (gst_video_rate_chain):
- Add some debug.
+2010-06-07 08:17:13 +0200 Martin Bisson <martin.bisson@gmail.com>
- * tests/check/elements/videorate.c: (GST_START_TEST),
- (videorate_suite):
- Added check for videorate changing caps handling. Closes #421834.
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add support for Y800 and Y16
+ Fixes bug #620441.
-2007-04-12 Michael Smith <msmith@fluendo.com>
+2010-06-07 08:16:01 +0200 Martin Bisson <martin.bisson@gmail.com>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Use scale functions to avoid overflow when calculating duration of
- vorbis buffers.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for Y800 and Y16
+ Fixes bug #620441.
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-06 16:46:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
- API: add gst_tag_freeform_string_to_utf8() (#405072).
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix log function printf format issue
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
- Use gst_tag_freeform_string_to_utf8() here.
+2010-06-05 18:14:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-12 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: stop jpeg typefinding once we found a SOF marker
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
- (gst_gdp_pay_sink_event):
- Make sure we set the IN_CAPS flag correctly.
+2010-06-05 18:05:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Get the IN_CAPS flag before we call functions that mess with the flags.
+ * tests/check/gst/typefindfunctions.c:
+ tests: fix memory leak in unit test
-2007-04-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-05-19 15:40:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
- gst_gdp_pay_chain, gst_gdp_pay_sink_event):
- Only stamp buffers with offset/offset_end right before they get
- pushed. This ensures offset continuity, which was not the case
- before as shown by
- gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: improve jpeg typefinder
+ Make jpeg typefinder check more than just the first two bytes
+ plus Exif or JFIF marker. This allows us to report MAXIMUM
+ probability in cases where there's no Exif or JFIF marker,
+ making typefinding stop early. Also extract width and height,
+ because we can.
-2007-04-06 Wim Taymans <wim@fluendo.com>
+2010-06-05 17:22:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybin.c: (add_sink),
- (gst_play_bin_change_state):
- Activate sync in playbin, we are ready to handle it for live streams.
+ * configure.ac:
+ * tests/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/gst/typefindfunctions.c:
+ * tests/files/Makefile.am:
+ * tests/files/partialframe.mjpeg:
+ tests: add small unit test for AC3 vs. JPEG typefinding issue
-2007-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-05 16:58:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/playbin.c:
- (test_sink_usage_video_only_stream), (playbin_suite):
- Add small test for stream-info-value-array code paths.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix AC-3 typefinding so that it actually checks for a second frame
+ Fix typo that made the AC-3 typefinder not actually check for a
+ second frame, but rather compare the sync point found to itself,
+ which resulted in the AC-3 typefinder reporting an overly optimistic
+ MAXIMUM or VERY_LIKELY probability when it found a possible frame
+ sync.
-2007-04-05 Wim Taymans <wim@fluendo.com>
+2010-06-05 12:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_skew_slaving):
- Don't try to create invalid calibration parameters by making the
- internal time go backwards, instead make external time go forward.
+ * gst/playback/gstscreenshot.c:
+ playbin2: improve screenshot code
+ Use appsrc and appsink in the screenshot code to make things nicer.
-2007-04-05 Wim Taymans <wim@fluendo.com>
+2010-06-05 11:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix documentation string
- * gst/playback/gstplaybasebin.c: (add_stream):
- Fix leak in add_stream(), when g_value_set_object() increases the
- refcount of streaminfo object. Fixes #426250.
+2010-06-05 11:05:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-03 David Schleef <ds@schleef.org>
+ * gst/playback/gstplaysink.c:
+ playsink: add convert-frame action signal
+ Add a convert-frame action signal.
+ Fixes #620279
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add a test pattern called "circular", which has concentric
- rings with varying radial frequency. The main purpose of this
- pattern is to test fidelity loss in a filter or scaler element.
- Notably, this pattern is scale invariant, and is optimally viewed
- with a width (and height) of 400.
+2010-06-05 11:02:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-03 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstscreenshot.c:
+ * gst/playback/gstscreenshot.h:
+ playbin2: move marshaller to screenshot
+ Move the marshaller for the convert_frame signal to the screenshot file in
+ preparation for moving it to playsink.
+ See #620279
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2010-06-05 10:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
- (deactivate_free_recursive):
- Decodebin2 doesn't unref pads it obtains in some occasions:
- - multiqueue src pads, when either connecting further or exposing
- - sink pads of new autoplugged elements
- - peer pads when recursively freeing elements
- Fixes #425455.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: move convert_frame to playsink
+ Move the convert_frame function to playsink and make it part of the API. This is
+ in preparation to add the convert_frame signal to playsink.
+ See #620279
-2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-05 10:31:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Add audio/x-raw-float support, now that audioconvert support
- non-native endianness floats.
+ * gst/playback/gstplaysink.c:
+ playsink: add property to get the last frame
+ Add a property to get the last video frame.
+ See #620279
-2007-03-30 Tim-Philipp Müller <tim at centricular dot net>
+2010-06-04 19:30:14 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- gstreamer-plugins-base.pc doesn't exist, it's
- gstreamer-plugins-base-0.10.pc.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Handle raw streams we don't want.
+ If a file contains raw streams (not requiring a decoder) that we do
+ not want (expose-all-streams == FALSE), we would previously consider
+ those of unknown-type (missing a decoder) ... whereas in fact it was just
+ because they don't need decoders.
+ This only applies if expose-all-streams is FALSE.
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-03 13:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler dot de>
- with some minor changes
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "baseaudiosink: Allocate and free the clock in NULL->READY and reverse"
+ This reverts commit cea2644ed86097aadedc9e8731e78a22ffc6246b.
+ Many audio sink assume that they can create a clock in
+ the instance init function and it will be there forever
+ and not be cleared by the state change functions.
- * gst-libs/gst/floatcast/floatcast.h:
- Use more efficient float endianness conversion functions that don't
- involve 2 function calls per value.
- * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
- (check_default), (audio_convert_prepare_context):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps), (make_lossless_changes):
- Support non-native endianness floats as input and output.
- Fixes #339838.
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- Add unit tests for the non-native endianness float conversions.
+2010-06-02 12:19:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: Allocate and free the clock in NULL->READY and reverse
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_base_init),
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_change_state),
- (gst_base_rtp_depayload_set_property),
- (gst_base_rtp_depayload_get_property):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Add Private structure.
- Bring element code to 2007.
- Parse clock-base caps param and use it when generating the
- newsegment.
- Reset variables before going to PAUSED.
- Fix some docs.
+2010-06-01 23:49:07 -0700 David Schleef <ds@schleef.org>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 17f89e5 to 47683c1
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_get_adapter):
- Add RTCP docs.
- Fix some more docs.
+2010-06-01 22:54:33 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
- (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
- (gst_rtcp_buffer_get_packet_count), (read_packet_header),
- (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
- (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
- (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
- (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
- (gst_rtcp_packet_sr_get_sender_info),
- (gst_rtcp_packet_sr_set_sender_info),
- (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
- (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
- (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
- (gst_rtcp_packet_sdes_get_chunk_count),
- (gst_rtcp_packet_sdes_first_chunk),
- (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
- (gst_rtcp_packet_sdes_first_item),
- (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
- (gst_rtcp_packet_bye_get_ssrc_count),
- (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
- (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
- (gst_rtcp_packet_bye_get_reason_len),
- (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Add new helper object for parsing and creating RTCP messages.
+ * common:
+ Automatic update of common submodule
+ From fd7ca04 to 17f89e5
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-01 13:00:22 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- PCM samples with width=8 must be always unsigned, no matter what
- depth they have.
+ * configure.ac:
+ * tests/examples/overlay/Makefile.am:
+ examples: get the right Qt moc binary to use via pkg-config
+ Should make us do the right thing in cases where both Qt3 and Qt4
+ are installed.
+ Fixes #620211.
-2007-03-29 Andy Wingo <wingo@pobox.com>
+2010-05-31 19:28:45 +1000 Jonathan Matthew <jonathan@d14n.org>
- * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
- perfect offsets also, not just timestamps.
+ * ext/gio/gstgiobasesink.c:
+ gio: map GIO NO_SPACE error to NO_SPACE_LEFT
+ Fixes bug #620140.
- * tests/check/elements/videorate.c (test_more): Test that given
- any incoming offsets, that videorate produces perfect offsets.
+2010-05-28 08:27:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ configure: Remove (now) useless check for cbrt
- * gst-libs/gst/riff/riff-ids.h:
- Add some more RIFF formats.
+2009-12-02 22:16:22 -0800 David Schleef <ds@schleef.org>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/interfaces/streamvolume.c:
+ interfaces: Use pow() instead of cbrt() for MSVC
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_default_clock_rate):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Fix fixed payload names and docs.
- Added method to get the default clock rates of fixed payload types.
- API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
-
-2007-03-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * tests/check/pipelines/.cvsignore:
- Add new vorbisdec test to cvsignore.
-
-2007-03-28 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
- (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
- (gst_base_audio_sink_set_property),
- (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
- (clock_convert_external), (gst_base_audio_sink_resample_slaving),
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
- (gst_base_audio_sink_async_play):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Store private stuff in GstBaseAudioSinkPrivate.
- Add configurable clock slaving modes property.
- API:: GstBaseAudioSink::slave-method property
- Some more latency reporting tweaks.
- Added skew based clock slaving correction and make it the default until
- the resampling method is more robust.
+2010-05-26 11:54:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-27 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 357b0db to fd7ca04
- * gst/audioconvert/audioconvert.c:
- Add docs to the integer pack functions and implement proper
- rounding. Before we had rounding towards negative infinity, i.e.
- always the smaller number was taken. Now we use natural rounding,
- i.e. rounding to the nearest integer and to the one with the largest
- absolute value for X.5. The old rounding introduced some minor
- distortions. Fixes #420079
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Fix one unit test that assumed the old rounding and added unit tests
- for checking signed/unsigned int16 <-> signed/unsigned int16 with
- depth 8, one for signed int16 <-> unsigned int16 and one for the new
- rounding from signed int32 to signed/unsigned int16.
-
-2007-03-27 Michael Smith <msmith@fluendo.com>
-
- * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
- (gst_audio_convert_transform_caps):
- Fix typo in debug line introduced recently, as pointed out on irc.
+2010-05-26 08:51:09 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-03-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Fix buffer offset_end when within tolerance.
+ This fixes issues if we then have downstream elements that operate
+ on offset/offset_end.
+ And add the expected timestamp in the debug logs
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- * tests/check/libs/tag.c: (GST_START_TEST):
- Make sure we parse floating-point numbers in vorbis comments
- correctly with either '.' or ',' as separator, no matter what
- the current locale is. Add unit test for this too.
+2010-05-24 11:27:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-26 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/fft/kiss_fft_f32.c:
+ * gst-libs/gst/fft/kiss_fft_f32.h:
+ * gst-libs/gst/fft/kiss_fft_f64.c:
+ * gst-libs/gst/fft/kiss_fft_f64.h:
+ * gst-libs/gst/fft/kiss_fft_s16.c:
+ * gst-libs/gst/fft/kiss_fft_s16.h:
+ * gst-libs/gst/fft/kiss_fft_s32.c:
+ * gst-libs/gst/fft/kiss_fft_s32.h:
+ * gst-libs/gst/fft/kiss_fftr_f32.c:
+ * gst-libs/gst/fft/kiss_fftr_f64.c:
+ * gst-libs/gst/fft/kiss_fftr_s16.c:
+ * gst-libs/gst/fft/kiss_fftr_s32.c:
+ fft: Merge kissfft 1.2.8
+ This reduces memory footprint for the FFT and adds
+ OpenMP support (but we don't use it).
- Patch by: René Stadler <mail at renestadler de>
+2010-05-22 10:05:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- When writing out floating-point numbers to vorbis comment tags, always
- use the same character as separator no matter what the current locale is
- (fixes #423051).
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate interlaced, chroma-site and color-matrix fields if necessary
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit tests for replaygain tags in vorbis comments (closes #423055).
+2010-05-22 10:02:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * gst/videorate/gstvideorate.c:
+ videorate: Use new string fixation function from core
- * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
- vorbis_handle_data_packet):
- Correctly set DURATION to generate a timestamp-continuous stream.
- One bug left at the end; see
- ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
- * tests/check/Makefile.am:
- * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
- Add a test to check this. Without the above patch this test fails.
+2010-05-22 09:48:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate color-matrix and chroma-site fields if necessary
- * gst-libs/gst/rtp/Makefile.am:
- The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
+2010-05-22 09:39:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-23 Michael Smith <msmith@fluendo.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate the interlaced field if necessary
+ Fixes bug #619310.
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
- (gst_video_rate_reset), (gst_video_rate_chain):
- If videorate changes caps, we can no longer use the old buffer
- (which may have a different size, incompatible with our caps).
- So don't do that; just duplicate the new frame more times.
+2010-05-22 08:55:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add IVF typefinder
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
- Remove playbin's override of the set_clock vmethod. It's irrelevant
- after Wim's commit on the 19th.
+2010-05-21 18:16:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-22 Wim Taymans <wim@fluendo.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: pass object to logging functions, use GST_DEBUG_FUNCPTR
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
- (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
- * ext/gnomevfs/gstgnomevfssrc.h:
- Don't cache file sizes. Fixes #341078.
+2010-05-20 15:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add basic descriptions for new WebM and VP8 types
- * gst/playback/gstplaybin.c: (add_sink):
- Use GST_PTR_FORMAT to log caps.
+2010-05-20 14:21:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix sizes again, this time for real
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2010-05-20 13:58:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_font):
- Special-case some more colour names that pango doesn't handle by
- default. Fixes #420578.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix size checks
-2007-03-20 Michael Smith <msmith@fluendo.com>
+2010-05-20 10:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- If we get a zero-sized input buffer, don't pass it to libvorbis, as
- that marks EOS internally. After that, libvorbis will buffer all
- input data, and encode none of it, eventually leading to memory
- exhaustion.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Drop all other Ogg VP8 header packets and make VP8 mapping check a bit more strict
-2007-03-19 Wim Taymans <wim@fluendo.com>
+2010-05-20 08:52:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (remove_fakesink):
- Don't post STATE_DIRTY anymore.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Some more minor adjustments for the VP8 Ogg mapping
- * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
- (gst_play_bin_change_state):
- Remove stream_time reset in seek handling, core does that now.
- Disable clocking for live pipelines by forcing a NULL clock to the
- complete pipeline, core is too smart now for our previous hack.
- We can always autoplug in PAUSED now.
+2010-05-19 21:35:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-17 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Update to the latest VP8 mapping
- * REQUIREMENTS: Update this file, change the formatting to make
- it more consistent, plus more machine readable.
+2010-05-10 05:53:22 +0200 Philip Jägenstedt <philipj@opera.com>
-2007-03-16 Michael Smith <msmith@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Detect WebM as video/webm
+ Refactor matroska_type_find into ebml_check_header and a new
+ matroska_type_find and webm_type_find.
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
- (strip_width_64), (append_with_other_format):
- Previous fix was too simplistic, and broke the tests. Use a better
- approach; only strip 64 from widths for integer audio.
+2010-05-14 13:31:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-16 Michael Smith <msmith@fluendo.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix granulepos->key granule calculation for Dirac
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
- (gst_audio_convert_transform_caps):
- We don't support 64 bit integer audio, so don't try to claim we can.
- Stops us producing caps don't match our template caps.
- Update comments.
+2010-05-14 11:02:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-15 Michael Smith <msmith@fluendo.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Add support for mapping specific granulepos to key granule mapping
- * gst/audioresample/gstaudioresample.c:
- (audioresample_check_discont), (audioresample_transform):
- Don't trigger discontinuities for very small imperfections; a filter
- flush will sound bad, and many plugins have rounding errors leading
- to these.
+2010-05-05 13:59:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ ogg: Implement Ogg VP8 mapping
- Patch by: Olivier Crete <olivier.crete@collabora.co.uk>
+2010-04-27 15:24:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- API: add "min-ptime" property to RTP base audio payloader.
- API: add gst_base_rtp_audio_payload_push().
- API: add gst_base_rtp_audio_payload_get_adapter().
- Fixes #415001
- Indentation/whitespace/documentation fixes.
-
-2007-03-14 Julien MOUTTE <julien@moutte.net>
-
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (audioresample_transform_size), (audioresample_do_output),
- (audioresample_transform), (audioresample_pushthrough): Handle
- discontinuous streams.
- * gst/audioresample/gstaudioresample.h:
- * tests/check/elements/audioresample.c:
- (test_discont_stream_instance), (GST_START_TEST),
- (audioresample_suite): Add a test for discontinuous streams.
- * win32/common/config.h: Updated.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for On2 VP8
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-05-19 16:17:19 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- Update translations from translation project.
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix a typo introduced by 9d753824.
+ video/x-raw-float => audio/x-raw-float. Fixes #619090.
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-05-18 08:45:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioresample/debug.h:
- * gst/audioresample/resample.c: (resample_init):
- Since I really am not interested in a debug line for each sample
- being processed, move the library's debugging to its own category,
- libaudioresample
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't put "raw" subtitle types in the raw caps for decodebin2
+ We handle them from the autoplug-continue signal, where the caps supported
+ by the subtitle sink or overlay are known already.
-2007-03-13 Michael Smith <msmith@fluendo.com>
+2010-05-15 21:15:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_handle_type_packet):
- Since the plugin doesn't support anything other than 4:2:0 right
- now, post an error and fail if we get something else. Won't matter
- until libtheora supports the other pixel formats, but hopefully
- that'll be soon...
+ * configure.ac:
+ configure: Use = instead of == in shell scripts for equality checks
-2007-03-10 Sebastien Moutte <sebastien@moutte.net>
+2010-05-14 18:23:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
- Use gst_guint64_to_gdouble for conversion.
- * win32/MANIFEST:
- Add new files to the win32 MANIFEST.
- * win32/common/libgstaudio.def:
- * win32/common/libgstpbutils.def:
- Add new exported functions.
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstdecodebin.dsp:
- * win32/vs6/libgstplaybin.dsp:
- Change the link to libgstpbutils.lib.
- * win32/vs6/libgstdecodebin2.dsp:
- Add a new project for decodebin2.
- * win32/vs6/libgstpbutils.dsp:
- Add a new project for pbutils.
+ * common:
+ Automatic update of common submodule
+ From 4d67bd6 to 357b0db
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-05-14 17:24:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Also accept partial dates with only year and month,
- like 1999-12-00 (fixes #410396 even more).
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Always take the object lock when accessing the caps
+ Fixes bug #618625.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit test for the above.
+2010-05-14 17:17:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Don't fail if subtitles are used but only audio is available and no visualizations
+ Instead simply disable displaying of the subtitles for now, as was
+ intended by that part of code...
+ Fixes bug #610866.
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add unit test for MPL2 subtitle format (#413799).
+2010-05-14 17:13:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Fix deadlock caused from an additional lock instead of unlock
+ Also improve debug output for the playsink lock.
- Patch by: Kamil Pawlowski <kamilpe gmail com>
+2010-05-13 12:16:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
- (gst_subparse_type_find):
- * gst/subparse/gstsubparse.h:
- * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
- * gst/subparse/mpl2parse.h:
- Add support for MPL2 subtitle format (#413799).
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use passthrough mode if width and height are not changed
+ It doesn't matter if the PAR changes or not, processing of every pixel
+ is only necessary when the width or height changes.
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2010-05-13 12:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- We require core CVS for the new buffer metadata copy functions.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: relax width and height constraints
+ Increase the acceptable video sizes from [16,4096] to [1, MAX].
+ See #618392
-2007-03-09 Wim Taymans <wim@fluendo.com>
+2010-05-13 08:05:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c:
- Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
- Fixes #414496.
+ * gst-libs/gst/video/video.c:
+ video: Use simple fraction multiplication functions instead of going through GValues
- Patch by: Alex Lancaster <alexl at users sourceforge net>
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a unit test for checking if the negotiation works as expected
- * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
- (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
- Improve adapter usage and comments.
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try harder to keep the DAR if possible
+ Fixes bug #371108.
- * ext/pango/gsttextrender.c: (gst_text_render_chain):
- * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
- * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
- Use new metadata copy function.
+2010-05-10 15:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
- Basetransform copied the metadata for us.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Log PAR and DAR of input and output caps when setting caps
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2010-05-10 14:52:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
- (gst_text_overlay_video_event):
- Some more logging. Only accept newsegment events in TIME format and
- send a WARNING message if they are not in TIME format.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Set input width/height if the output caps don't have any width or height
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
- (gst_sub_parse_chain), (gst_sub_parse_sink_event):
- * gst/subparse/gstsubparse.h:
- No need to allocate GstSegment structure dynamically, just put it
- into the instance structure; ignore newsegment events in BYTE
- format and in particular don't let it overwrite our saved TIME
- segment from the last seek.
+2010-05-10 13:01:44 +0200 Andoni Morales <ylatuya@gmail.com>
-2007-03-09 Michael Smith <msmith@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try to keep DAR when scaling
+ Fixes bug #371108.
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
- Replace AC3 typefinder with one that isn't terrible, and actually
- works usefully.
+2010-05-10 19:09:28 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ basertpaudiopayload: Add extra frame for non-complete frame lengths
+ Some payloaders like rtpg729pay can add a shorter frame at the end of a
+ RTP packet. We need to count it like a full frame for timestamps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=618324
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- fix error category and translatable string
-
+2010-05-10 18:53:29 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ basertpaudiopayload: Set duration on buffers
+ Set the duration of the buffers from their size
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Fix up utils => pbutils here too.
+2010-05-11 16:12:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate PAR to 1/1 if possible
- * gst/subparse/gstsubparse.c: (handle_buffer):
- Break out of loop in chain function as soon as possible if we get
- a non-OK flow return.
+2010-05-11 10:07:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0
- * tests/check/elements/alsa.c: (GST_START_TEST):
- Unref the mixer if the state change fails too (if the
- alsa devices are inaccessible, for example)
+2010-05-10 12:44:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * sys/ximage/ximagesink.c:
+ ximagesink: Check if the X context is allocated before using it
+ It should be allocated at these places already or the state changes
+ would have failed... but better add an additional check here.
- * tests/check/Makefile.am:
- Don't test libvisual elements in the states check, because libvisual
- seems to leak internally.
+2010-05-10 12:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Re-enable the alsa and states tests now that there's new suppressions
- in gst.supp.
+ * sys/ximage/ximagesink.c:
+ ximagesink: Post an error message on the bus if no supported pixmap formats can be found
+ Might fix bug #615851.
- * tests/check/elements/alsa.c: (GST_START_TEST):
- Don't leak the alsamixer we instantiated.
+2010-05-07 19:49:57 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace : whooops
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_change_state), (gst_ximagesink_reset),
- (gst_ximagesink_finalize):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
- (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
- Move some cleanup stuff from the state change handler into a _reset()
- function that can be called from _finalize(). This ensures that things
- get freed even if (for some reason) the NULL->READY state transition
- fails in the parent class.
- Even if a parent state change fails, process our downward state change
- logic instead of bailing out early.
- Free the correct xcontext pointer in ximagesink's xcontext_clear.
+2010-05-07 19:21:13 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: more minor cleanups
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Extra log line.
+2010-05-07 17:16:28 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
- * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
- Use pango_font_description_set_family_static instead of
- pango_font_description_set_family to save a string copy (it was
- leaking due to the strdup anyway)
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: speedup caps transformation
+ * don't re-create our possible caps every single time, just use the
+ template caps.
+ * don't intersect the caps against the template, basetransform has already
+ done that for us.
+ 62% speedup of _transform_caps() (instruction calls, measured with callgrind)
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
- Chain up in finalize.
+2010-05-07 12:19:25 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add the 'expose-all-streams' property from decodebin2
+ API: expose-all-streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
- * gst-libs/gst/interfaces/mixertrack.c:
- (gst_mixer_track_class_init), (gst_mixer_track_get_property),
- (gst_mixer_track_set_property):
- API: add "untranslated-label" property which should be set by
- implementations at construct time (#414645).
+2010-05-06 18:50:51 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Set "untranslated-label" when constructing mixer track objects.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Add a property to not expose/decode all streams
+ API : expose-all-streams
+ If disabled:
+ * only the streams that CAN be decoded and match the final caps will have a
+ decoder plugged in and be exposed.
+ * the streams that COULD HAVE BEEN decoded but do not match the finals caps
+ will not have a decoder plugged in and will not be exposed.
+ If no decoder is available to decode a certain stream, then the missing element
+ message will still be emitted regardless of the value of the property.
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
+
+2010-05-06 17:47:12 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
- Unit test to check the above.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: rename are_raw_caps to are_final_caps, correct comment
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
-2007-03-07 Wim Taymans <wim@fluendo.com>
+2010-05-07 17:16:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
- Fix confusing debug message.
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ sdpmessage: add new TIAS bandwidth modifier
+ Add TIAS modifier as specified in RFC 3890.
+ Do some whitespace fixes.
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+2010-05-07 00:10:22 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-plugins-base.doap:
- update doap file with new version
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: disambigue comment due to popular demand
+ Write "target depth" instead of "our depth" or previous ambigous "out depth".
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+2010-05-06 15:40:34 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- Back to CVS
+ * gst/playback/gstplaysink.c:
+ playsink: disconnect signals in some more cleanup cases
-=== release 0.10.12 ===
+2010-05-06 13:10:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't seek when no current chain
+ Avoid a crash when we try to seek when there is no current chain.
- * configure.ac:
- releasing 0.10.12, "Zombie Horde"
+2010-05-06 12:21:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: ignore the skeleton start time
+ Ignore the skeleton start time as it is usually wrong for live streams
+ and we have the needed logic to calculate it anyway.
- * configure.ac:
- Bump version to 0.10.11.4 pre-release
+2010-05-06 12:06:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-06 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: wait for headers before exposing chains
+ Wait until we have all the stream headers before we start exposing the streams
+ of a chain.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- Fix regression that made GStreamer skip the first samples of audio.
- Fixes #414684.
+2010-05-06 10:56:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: use index to estimate bitrate
+ When we have an index, use it to much more accurately estimate the total stream
+ bitrate.
- * configure.ac:
- Bump version to 0.10.11.3 pre-release
+2010-05-06 11:34:53 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ docs: be more firendly to gtk-doc limitted parsing capabilities
- * po/POTFILES.in:
- Update paths for the rename from utils to pbutils to fix the build.
+2010-05-06 09:42:02 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ docs: fix wrong doc markup
- * gst-libs/gst/pbutils/Makefile.am:
- Change directory to install headers in from gst/utils to gst/pbutils
- as well.
+2010-05-06 09:17:33 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: use can_intersect to avoid a caps copy
- * configure.ac:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/interfaces/mixer.c:
- * gst-libs/gst/pbutils/Makefile.am:
- * gst-libs/gst/pbutils/descriptions.c:
- (gst_pb_utils_get_source_description),
- (gst_pb_utils_get_sink_description),
- (gst_pb_utils_get_decoder_description),
- (gst_pb_utils_get_encoder_description),
- (gst_pb_utils_get_element_description),
- (gst_pb_utils_add_codec_description_to_tag_list),
- (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
- * gst-libs/gst/pbutils/descriptions.h:
- * gst-libs/gst/pbutils/install-plugins.c:
- * gst-libs/gst/pbutils/install-plugins.h:
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_uri_source_message_new),
- (gst_missing_uri_sink_message_new),
- (gst_missing_element_message_new),
- (gst_missing_decoder_message_new),
- (gst_missing_encoder_message_new),
- (gst_missing_plugin_message_get_description):
- * gst-libs/gst/pbutils/missing-plugins.h:
- * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
- * gst-libs/gst/pbutils/pbutils.h:
- * gst-libs/gst/utils/Makefile.am:
- * gst-libs/gst/utils/base-utils.c:
- * gst-libs/gst/utils/base-utils.h:
- * gst-libs/gst/utils/descriptions.c:
- * gst-libs/gst/utils/descriptions.h:
- * gst-libs/gst/utils/install-plugins.c:
- * gst-libs/gst/utils/install-plugins.h:
- * gst-libs/gst/utils/missing-plugins.c:
- * gst-libs/gst/utils/missing-plugins.h:
- * gst-plugins-base.spec.in:
- * gst/playback/Makefile.am:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstdecodebin2.c:
- * gst/playback/gstplaybasebin.c: (setup_subtitle),
- (gen_source_element):
- * gst/playback/gstplaybin.c: (plugin_init):
- * tests/check/Makefile.am:
- * tests/check/libs/pbutils.c: (GST_START_TEST),
- (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
- * tests/check/libs/utils.c:
- rename utils to pbutils
+2010-05-06 09:14:25 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-02 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: trucate own caps, instead of copying and using the first only
+ We got the caps from an intersect, it is our own, hence we can truncate it.
+ Besides gst-indent has chooses to line-up all caps in one line again :/.
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-decodebin2.xml:
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
- Add documentation for decodebin2 that indicates that the API
- is still unstable.
+2010-05-06 09:12:32 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-01 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstdecodebin.c:
+ decodebin: use can_intersect to avoid a caps copy
- * configure.ac:
- Update to 0.10.11.2 (0.10.12 pre-release)
+2010-05-06 09:11:17 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * ext/libvisual/visual.c:
+ libvisual: trucate own caps, instead of copying and using the first only
+ We got the caps from an intersect, it is our own, hence we can truncate it.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- base time is irrelevant here.
+2010-05-06 08:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/gstvorbisdeclib.c:
+ * ext/vorbis/gstvorbisdeclib.h:
+ vorbis: have a copy_sample func as a func pointer
+ Make some more variants for copy_sample funcs and use them via function pointer.
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
- * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
- Improve debugging.
+2010-05-06 08:16:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_query), (gst_base_audio_sink_event),
- (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
- Improve latency and clock slaving calculations.
- Improve slave clock calibration.
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: fix typo in comment
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full):
- When we are asked to render N sample to 0 bytes, return N.
+2010-05-06 08:15:16 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: use gst_caps_can_intersect() more
+ In place where we just need to know whether caps intersect, we can use this
+ quicker function.
- * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
- (gst_alsasink_write), (gst_alsasink_reset):
- * ext/alsa/gstalsasink.h:
- Remove unused dispose function.
- Rename lock to not interfere with alsasrc lock.
+2010-04-15 13:09:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
- (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
- (gst_alsasrc_read), (gst_alsasrc_reset):
- * ext/alsa/gstalsasrc.h:
- Implement finalize function.
- Use lock to protect alsa access.
- Implement _reset.
- Fine tune sw params.
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/position-formats.c:
+ examples: add a test for difference position formats
+ The test runs position and duration queries on the pipeline in all formats.
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-04-15 13:08:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Convert to new AG_GST style.
+ * tests/icles/audio-trickplay.c:
+ example: update status (adder is fixed now)
-2007-02-28 Wim Taymans <wim@fluendo.com>
+2010-04-15 13:08:01 +0300 Stefan Kost <ensonic@users.sf.net>
- Patch by: Ed Catmur <ed at catmur dot co dot uk>
+ * tests/icles/playbin-text.c:
+ example: make app static
- * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
- (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
- Fix race condition when rapidly switching visualisations in playbin.
- Fixes #401029.
+2010-05-05 13:25:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-28 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: printf format fixes
- * tests/check/Makefile.am:
- Include local stuff before system installed things in LDFLAGS and
- CFLAGS.
+2010-05-04 15:32:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: weekday and month names in RTSP date string should be in C locale
+ Create date string using C locale weekday and month names.
+ Fixes #617636.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
- Improve debugging.
+2010-05-04 17:54:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add all qtdemux types to downloadable types
+ Add all the media types that qtdemux can handle to the list of downloadable
+ types.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
- (gst_v4lsrc_fixate), (gst_v4lsrc_query):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
- Fix duration and timestamping, taking latency into account.
- Implement latency query.
+2010-05-04 17:38:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: handle corrupt indexes
+ Make sure we handle and receover from corrupt indexes.
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
- (gst_audio_clock_new):
- Fix clock name.
+2010-05-04 15:47:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_init), (gst_base_audio_sink_query):
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
- (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
- (gst_base_audio_src_create):
- Improve latency query code.
- Use proper clock names.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix EOS check
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-05-04 13:51:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/generic/states.c: (GST_START_TEST):
- Copy the states.c test from core again
- * tests/check/Makefile.am:
- ignore cdio and cdparanoiasrc
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoild division by 0
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2010-05-04 13:50:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
- (double_hq), (audio_convert_get_func_index), (check_default),
- (audio_convert_prepare_context), (audio_convert_convert):
- Also make valgrind happy and avoid copying data in some cases.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: cleanup unused defines
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2010-05-04 13:36:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
- (double_hq), (audio_convert_get_func_index),
- (audio_convert_prepare_context), (audio_convert_convert):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
- (gst_audio_convert_transform_caps):
- * tests/check/elements/audioconvert.c: (GST_START_TEST),
- (audioconvert_suite):
- Don't run inplace if that overwrites source data as we go. Add more
- tests. Fixes #339837 even more.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use the index in push mode when we can
+ When seeking in push mode, try to use the index first before we use the bitrate
+ estimation.
-2007-02-27 Julien MOUTTE <julien@moutte.net>
+2010-05-04 13:05:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
- (msg_segment_done): Fix various seeking bugs (Slider was not
- updating when doing a non flushing seek, Reverse playback
- on segment seek was wrong).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use skeleton duration when possible
-2007-02-26 Wim Taymans <wim@fluendo.com>
+2010-05-04 13:02:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (stop_seek):
- When we stop scrubbing, don't leave the pipeline PLAYING when we
- requested a PAUSED state.
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggstream: parse duration from 3.3 skeleton
-2007-02-25 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-02 11:16:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler de>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: more index parsing work
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Parse date strings in vorbis comments that have an invalid (zero)
- month or day (#410396).
+2010-03-01 13:50:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/libs/tag.c: (GST_START_TEST):
- Test case for the above.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: clean up fishead/fisbone parsing
+ Remove some redundant code for parsing fishead streams. Actually use the data we
+ parsed (mostly start_time).
-2007-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2010-05-04 11:19:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Loïc Minier <lool+gnome at via ecp fr>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: implement seek in push mode
+ Refactor start time collection code.
+ When we receive a flush_stop, resync to the new start time and push out a new
+ segment event.
- * configure.ac:
- * ext/alsa/Makefile.am:
- * gst/audiotestsrc/Makefile.am:
- Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
+2010-05-03 16:52:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: make event handling better
+ Explicitly handle FLUSH events and resync on FLUSH_STOP.
+ Make send_event return a boolean.
+ Use more performant send_event function to forward events.
- * gst/playback/gstplaybin.c:
- Improve docs: point out that the application needs to assist playbin
- with buffering.
+2010-04-30 18:37:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: implement seeking in pushmode
+ Convert seek requests to bytes using the bitrate and forward them upstream. Does
+ not quite work because the flushing and resyncing is not implemented yet.
- * gst-libs/gst/utils/install-plugins.c:
- * gst-libs/gst/utils/missing-plugins.c:
- * tests/check/libs/utils.c: (missing_msg_check_getters):
- Change GStreamer marker prefix in detail string from 'gstreamer.net'
- to just 'gstreamer'. Document the caps string component of the
- decoder/encoder detail a bit better, since not everyone will be
- familiar with the GStreamer media type/caps system (but they better
- enjoy nested itemized lists).
+2010-04-30 18:03:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: refactor for seeking in pushmode
+ refactor the code a little to prepare for seeking in push mode
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
- Fix copying of GstNetBuffer (would crash before, or at least lead to
- invalid memory access, #410772), for now by copying the GstBuffer copy
- code from the core over here so we can copy the GstBuffer fields on a
- provided buffer instance (of type GstNetBuffer in this case). Would be
- better to fix this with some support by the core though (and in the long
- run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
+2010-05-03 12:46:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/check/Makefile.am:
- Enable unit test for GstNetBuffer.
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds new geo-location mappings
+ Adds GST_TAG_GEO_LOCATION_MOVEMENT_SPEED,
+ GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION and
+ GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION to xmp
+ mappings.
+ Tests included.
-2007-02-22 Andy Wingo <wingo@pobox.com>
+2010-04-26 22:08:41 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_init): Disable pull-mode activation until we
- figure out how to make audio sinks go to PLAYING.
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds xmp mappings for device tags
+ Adds xmp mappings for GST_TAG_DEVICE_MANUFACTURER and
+ GST_TAG_DEVICE_MODEL. Also adds tests for it.
-2007-02-22 Stefan Kost <ensonic@users.sf.net>
+2010-04-30 19:56:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
- (double_hq), (audio_convert_get_func_index),
- (audio_convert_prepare_context), (audio_convert_convert):
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
- (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
- * gst/audioconvert/gstchannelmix.h:
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Add float as an intermediate format, as well as float mixing. Enable
- test that was failing before. Fixes #339837
+ * gst-libs/gst/rtsp/gstrtspbase64.c:
+ * gst-libs/gst/rtsp/gstrtspbase64.h:
+ rtsp: deprecate remaining base64 function now that we depend on GLib 2.20
+ API: deprecate gst_rtsp_base64_decode_ip(), use g_base64_decode_inplace() instead
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+2010-04-30 19:37:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek):
- Undo the previous commit: -1 as a stop time implies that the stop
- time is the end of file, clearing any previously configured segment.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: ensure writable metadata
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+2010-04-30 17:41:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek):
- Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: use bitrate to estimate length in pushmode
+ Parse the bitrate from the various streams.
+ Use the bitrate and the upstream length in bytes to estimate the total stream
+ duration in push mode.
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+2010-04-30 14:07:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_process_int16),
- (volume_process_int16_clamp), (volume_set_caps):
- Unbreak volume, value remains gint.
+ * configure.ac:
+ * ext/gio/gstgiobasesrc.c:
+ Bump GLib requirement to 2.20
+ See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+2010-04-30 13:36:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_choose_func),
- (volume_update_real_volume), (gst_volume_set_volume),
- (gst_volume_init), (volume_process_double), (volume_process_float),
- (volume_process_int16), (volume_process_int16_clamp),
- (volume_set_caps), (volume_transform_ip), (volume_update_volume):
- * gst/volume/gstvolume.h:
- Extend float audio support (double) and some int->uint cleanups.
+ * gst/playback/gstplaybin2.c:
+ playbin2: forward duration query duration during group switch if no cached duration
+ ... such as during first group setup.
+ Fixes #616396.
-2007-02-20 Edward Hervey <edward@fluendo.com>
+2010-04-02 16:37:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
- (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
- (sort_end_pads), (gst_decode_group_expose),
- (gst_decode_group_hide):
- Don't free groups from the streaming threads. Just put them aside and
- free them in dispose.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: only EOS when all streams are EOS
-2007-02-20 Edward Hervey <edward@fluendo.com>
+2010-04-02 16:36:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (connect_element),
- (pad_added_group_cb), (gst_decode_group_check_if_blocked),
- (sort_end_pads), (gst_decode_group_expose):
- Handle dynamic pads within groups.
- Sort pads before exposing them in order to make playbin happy.
- There still is a race with the multiqueue filling up. This should be
- solved separately.
- Fixes #398721
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix debug message
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-30 08:45:43 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/utils/base-utils.c:
- * gst-libs/gst/utils/descriptions.c:
- * gst-libs/gst/utils/install-plugins.c:
- * gst-libs/gst/utils/missing-plugins.c:
- Some more docs (and descriptions for two subtitle formats).
+ * tests/icles/audio-trickplay.c:
+ test: fix copy and paste error of variable name
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-18 20:46:37 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/audio.c:
- Fix documentation.
+ * gst/adder/gstadder.c:
+ adder: only accept seek-types none and set
+ Previously we were also acting on cur and end, but treating them like none.
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-14 23:31:20 +0300 Stefan Kost <ensonic@users.sf.net>
- Patch by: Yves Lefebvre <ivanohe abacom com>
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: rework timestamping
+ Adder was using always incrementing timestamps. Seeking was done by setting the
+ position in the newsegment event. This was failing when doing segmented seeks
+ with rate<0.0, as offset (and thus timestamp) would go below 0.
+ Now we take both cur and end from the seek event. We construct newsegment events
+ depending including cur and end from the seek event. We set position to the
+ start of the segment. Timestamp is set to start or end of segment depending on
+ rate. Offset is recalculated.
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
- Don't leak caps. Fixes #408278.
+2010-04-26 17:30:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-15 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaysink.c:
+ playsink: Add support for deinterlacing
+ This is disabled by default and can be enabled with the
+ deinterlace flag.
+ Fixes bug #547603.
- * ext/cdparanoia/gstcdparanoiasrc.h:
- * ext/ogg/gstoggdemux.h:
- * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
- (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
- (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
- * gst-libs/gst/audio/audio.h:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/interfaces/videoorientation.h:
- * gst/adder/gstadder.h:
- More docs coverage and some ChangeLog surgery (add missing names)
+2010-04-26 11:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-15 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ playbin2: Add flag for enabling/disabling automatic deinterlacing
- * sys/ximage/ximagesink.c:
- (gst_ximagesink_calculate_pixel_aspect_ratio):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio):
- Small constifications.
+2010-04-26 11:11:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-15 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplay-enum.c:
+ playbin: Use g_once_init_{enter,leave} instead of GOnce for enum/flag registration
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
- (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
- (gst_base_audio_sink_async_play),
- (gst_base_audio_sink_change_state):
- Answer latency query.
- Use configured latency when syncing.
- Fix clock slaving.
+2010-04-23 17:01:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
- (gst_base_audio_src_query), (gst_base_audio_src_change_state):
- Fix possible memleak.
- Implement latency query.
- Small cleanups.
+ * gst/ffmpegcolorspace/Makefile.am:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ ffmpegcolorspace: Use GST_BOILERPLATE and use GstVideoFilter as base class
+ This gives automatic QoS handling.
-2007-02-15 Wim Taymans <wim@fluendo.com>
+2010-04-23 16:24:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
- Ignore errors in reset, these are not fatal. They also grab the element
- lock which is already taking when this function is called. Fixes
- #405451.
+ * gst/playback/gstplaysink.c:
+ playsink: Correctly reconfigure the video chain when switching from a subtitle to a non-subtitle file
+ Fixes bug #616422.
-2007-02-13 Stefan Kost <ensonic@users.sf.net>
+2010-04-23 16:08:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Remove 'tests/examples/xerror/Makefile' from output files again.
+ * gst/playback/gstplaybin2.c:
+ playbin2: If a text sink is provided, let subtitle parsing be done by decodebin2 if required
+ This way subtitle sinks only get buffers in the format that they
+ understand, i.e. raw parsed text in most cases.
+ Fixes bug #614942.
-2007-02-13 Stefan Kost <ensonic@users.sf.net>
+2010-04-23 15:30:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * docs/plugins/Makefile.am:
- Also crossref against gst-plugins-base-libs.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Set subtitle encoding on the decodebins again
-2007-02-12 Stefan Kost <ensonic@users.sf.net>
+2010-04-23 15:22:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Add crossreferences to glib/gobject/gstream docs.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: API: Add GST_VIDEO_FORMAT_v308 for packed 4:4:4 YUV
- * gst-libs/gst/audio/audio.h:
- Source formatting.
+2010-04-23 15:14:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
- Add own debug category.
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Some random cleanup
-2007-02-12 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-23 15:06:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler de>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for Y444, Y42B and Y41B
- * gst-libs/gst/tag/gstvorbistag.c:
- Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
- (#403597).
+2010-04-23 14:42:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Reorder template caps by the amount of information contained in the color formats
- * gst/playback/gstplaybasebin.c: (setup_source):
- When we have external subtitles and wait for the subtitle decodebin
- to get up and running, we set up a (sync) bus handler for the
- subtitle decodebin, so we can stop waiting when it posts an error
- message. However, we should do that before we set the subtitle
- decodebin's state to playing, otherwise things are racy and we might
- miss error messages posted before we had a chance to set up the bus.
- This should finally fix totem hanging on .txt pseudo-subtitle files.
-
-2007-02-10 Sébastien Moutte <sebastien at moutte dot net>
+2010-04-22 15:46:17 -0400 Joshua M. Doe <joshua.doe@us.army.mil>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
- Use gst_gdouble_to_guint64 for conversions.
- * win32/common/config.h.in:
- Add a define for GST_INSTALL_PLUGINS_HELPER
- * win32/common/libgstaudio.def:
- * win32/common/libgstcdda.def:
- * win32/common/libgstnetbuffer.def:
- * win32/common/libgstrtp.def:
- * win32/common/libgutils.def:
- Add new exported functions.
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstdecodebin.dsp:
- * win32/vs6/libgstnetbuffer.dsp:
- * win32/vs6/libgstplaybin.dsp:
- * win32/vs6/libgstrtp.dsp:
- * win32/vs6/libgstvorbis.dsp:
- * win32/vs6/libgstcdda.dsp:
- * win32/vs6/libgstgdp.dsp:
- * win32/vs6/libgstutils.dsp:
- Update and add new project files.
+ * gst/videorate/gstvideorate.c:
+ videorate: add support for video/x-raw-gray
-2007-02-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-29 15:05:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
- (subrip_remove_unhandled_tags), (parse_subrip):
- For SubRip (.srt) subtitles, ignore all markup tags we don't
- handle (like font tags, for example).
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ docs: remove references to and introspection data of plugins that no longer exist
+ Some plugins (decodebin2, playbin) have been renamed or merged
+ into different plugins (uridecodebin, playback).
- * tests/check/elements/subparse.c:
- Add test for this.
+2010-04-29 15:02:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development.
- * gst/playback/gstdecodebin.c: (add_fakesink),
- (gst_decode_bin_change_state):
- * gst/playback/gstdecodebin2.c: (add_fakesink),
- (gst_decode_bin_change_state):
- Don't error out if there is no fakesink in the NULL to READY state
- change, since when decodebin is re-used, we're only adding the
- fakesink element in READY to PAUSED.
+=== release 0.10.29 ===
- * tests/check/elements/decodebin.c:
- (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
- (decodebin_suite):
- Minimal unit test to make sure we can use the same decodebin
- instance twice (at least with audiotestsrc input).
+2010-04-28 02:16:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.29
- * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
- Try to get devic-name from device string first, and from handle only
- as fallback (seems to yield better results and is more robust
- against buggy probing code on the application side).
+2010-04-28 01:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-08 Tim-Philipp Müller <tim at centricular dot net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- Based on patch by: Julien Puydt <julien.puydt at laposte net>
+2010-04-25 23:14:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
- (gst_alsa_find_device_name):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
- Improve device-name detection a bit, especially in the case where
- the device is not actually open (#405020, #405024). Move common code
- into gstalsa.c instead of duplicating it.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.28.3 pre-release
-2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-20 17:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
+ * gst-plugins-base.doap:
+ doap: update repository info from cvs->git and maintainers
-2007-02-06 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_interface_supported),
- (gst_xvimagesink_probe_get_properties),
- (gst_xvimagesink_probe_probe_property),
- (gst_xvimagesink_probe_needs_probe),
- (gst_xvimagesink_probe_get_values),
- (gst_xvimagesink_property_probe_interface_init),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init),
- (gst_xvimagesink_get_type):
- * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
- for XVAdaptors so that one can choose the adaptor to use with
- gstreamer-properties.
-
-2007-02-06 Stefan Kost <ensonic@users.sf.net>
+2010-04-23 14:39:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Also mention that a conversion from double to float is suboptimal still.
+ * common:
+ Automatic update of common submodule
+ From fc85867 to 4d67bd6
-2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-22 20:58:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init), (gst_audio_filter_change_state):
- Clear our formats structure and free the caps contained in it when
- shutting down.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y41B->Y444 conversion
+ ...which is the intermediate conversion for conversion to all
+ other formats.
+ Fixes bug #616545.
-2007-02-05 Andy Wingo <wingo@pobox.com>
+2010-04-16 20:03:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_callback): Update basesink->offset so that we
- pull monotonically increasing offsets instead of, um, seeking back
- to 0 each time. Fixes alsasrc ! alsasink!
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Don't leak the input buffer in error cases
+ Fixes bug #615572.
-2007-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-29 12:53:11 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/videoscale/gstvideoscale.c:
- A width and height of 1 makes us crash, so increase minimum size to
- 2x2 pixels until someone feels like fixing this (#404512).
+ * ext/ogg/gstoggmux.c:
+ docs: fix typo in link name
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-15 12:59:53 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
- Add small test to make sure request pads are cleaned up properly
- even if oggmux never changes state out of NULL.
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: gracefully handle ximagesink>xwindow == NULL
+ Expose could be called before we have set the xwindow. Handle this gracefully
+ like we do in image_put.
+ Fixes #615789
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-15 11:44:49 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/libs/utils.c: (GST_START_TEST):
- Fix unit test. Turns out things work much better when you
- NULL-terminate string arrays. Should make p5 build bot happy again.
+ * sys/ximage/ximagesink.c:
+ ximagesink: refactor _update_geometry()
+ Refactor like in xvimagesink. Remove the extra parameter and adjust the assert check.
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-15 07:18:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_base_init),
- (gst_audio_filter_template_class_init),
- (gst_audio_filter_template_init),
- (gst_audio_filter_template_set_property),
- (gst_audio_filter_template_get_property),
- (gst_audio_filter_template_setup),
- (gst_audio_filter_template_filter),
- (gst_audio_filter_template_filter_inplace), (plugin_init):
- Oops, forgot to commit fixed-up example.
+ * configure.ac:
+ configure: Drop -Wcast-align
+ Commit message copied from core's commit from Benjamin Otte:
+ 246f5dba96a5b50bb74621af67b30942cca72af5
+ Apparently gcc warns that GstMiniObject is not castable to
+ GstEvent/Message/Buffer due to them containing 64bit variables, even
+ though ARM hackers claim that those only need 4byte alignment. And as
+ long as gcc behaves that way, this warning is not very useful.
+ So we'll remove the warning until this problem is fixed.
+ Fixes #615698
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-14 14:13:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
- (gst_audio_filter_class_init), (gst_audio_filter_init),
- (gst_audio_filter_set_caps),
- (gst_audio_filter_class_add_pad_templates):
- * gst-libs/gst/audio/gstaudiofilter.h:
- Port GstAudioFilter to 0.10. This change technically breaks
- API and ABI (and thus also every library developer's heart),
- but seems justifiable on the grounds that the base class was
- completely unusable before (ie. would crash immediately when
- actually used). Fixes #403963 (and eventually also #403572).
- Also document all of this a bit.
+ * configure.ac:
+ * gst-libs/gst/tag/lang-tables.dat:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.28.2 pre-release
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-14 13:50:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/utils/install-plugins.c:
- (gst_install_plugins_spawn_child):
- * tests/check/libs/utils.c:
- (test_base_utils_install_plugins_do_callout):
- Lowering log level to see why things fail on the p5 build bot;
- fix some typos in unit test messages.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-13 16:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/libs/utils.c:
- (test_base_utils_install_plugins_do_callout):
- Don't hard-code temp directory for test helper; use GLib functions
- to write out file and do error checking etc.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: init geometry when setting new xid
+ Don't rely on expose event to query geomentry after new xid is set.
+ Fixes #615647.
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/utils/Makefile.am:
- * gst-libs/gst/utils/base-utils.h:
- * gst-libs/gst/utils/install-plugins.c:
- (gst_install_plugins_context_set_xid),
- (gst_install_plugins_context_new),
- (gst_install_plugins_context_free),
- (gst_install_plugins_get_helper),
- (gst_install_plugins_spawn_child),
- (gst_install_plugins_return_from_status),
- (gst_install_plugins_installer_exited),
- (gst_install_plugins_async), (gst_install_plugins_sync),
- (gst_install_plugins_return_get_name),
- (gst_install_plugins_installation_in_progress):
- * gst-libs/gst/utils/install-plugins.h:
- API: add API for applications to initiate installation of missing
- plugins, ie. gst_install_plugins_async() primarily.
- Based on libgimme-codec by Ryan Lortie.
+2010-04-14 13:43:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Add --with-install-plugins-helper configure option so distros can specify
- the path of the helper script or program to call when plugin installation
- is requested (distros: please do any argument munging in this helper
- script instead of patching GStreamer to pass arguments differently
- to another program directly).
+ * gst/audioconvert/Makefile.am:
+ * tests/examples/app/Makefile.am:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/gio/Makefile.am:
+ * tests/examples/volume/Makefile.am:
+ * tests/old/examples/switch/Makefile.am:
+ build: use LDADD instead of LDFLAGS to specify libs to link to when building executables
+ Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
+ This should make sure arguments are passed to the linker in the right
+ order, and makes LDFLAGS usable again.
+ Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
+ Fixes #615697.
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Build and document new API.
+2010-04-12 14:02:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/libs/utils.c: (result_cb),
- (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
- (libgstbaseutils_suite):
- Some simple checks for the new API.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add channels and rate to ADTS caps if we can
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-12 13:33:18 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (test_float_conversion):
- Add small test for 32bit float <=> 64bit float conversion (works
- only one way so far, 32=>64 produces structured noise).
+ * gst/typefind/Makefile.am:
+ * gst/typefind/gstaacutil.c:
+ * gst/typefind/gstaacutil.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add AAC level to ADTS caps
+ This adds code to calculate the level for a given AAC stream and export
+ it in the stream caps. For AAC LC streams, the level is calculated
+ according to the definition under the AAC Profile. For other streams,
+ the definition under the Main Profile is used.
+ HE-AAC support is still to be done, and is dependent on detecting the
+ presence of SBR and PS in the stream.
+ Level is added as a field of type string because that's the way it's
+ done in H.264 caps as well. There are only a few possible levels, so
+ not using a numerical type is not too painful in this case, and
+ consistency is nice.
+ Fixes #613589.
+
+2010-03-10 13:32:53 +0000 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add AAC profile to ADTS caps
+ This looks at the AAC profile for ADTS streams and adds the profile as a
+ string in the corresponding caps.
+ Profile is the actual profile, base-profile denotes the minimum codec
+ requirements to decode this stream. In this case they're always the
+ same, but they may differ e.g. in case of certain HE-AAC streams that
+ can be partially decoded by LC decoders (with loss of quality of course)
+ if no suitable HE-AAC decoder is available.
+ Fixes #612312.
+
+2010-04-11 22:58:15 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/audioconvert/gstaudioconvert.c:
- (set_structure_widths_32_and_64), (make_lossless_changes):
- We don't support floats with a width of 40, 48 or 56 bits.
+ * gst/adder/gstadder.c:
+ adder: add support for negative playback rates
+ Decrement sample counter when playing backwards. Set proper segment when playing
+ backwards (0..cur instead or cur..-1). Add more logging and fix a format string.
-2007-02-02 Stefan Kost <ensonic@users.sf.net>
+2010-03-26 19:00:47 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float), (double),
- (audio_convert_get_func_index):
- * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
- (make_lossless_changes):
- Support for 64-bit float audio in audioconvert (#339837)
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: use ptime-multiple
+ Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>
+ Fixes #613248
-2007-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-09 16:06:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Holger Wansing <linux wansing-online de>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ audiopayload: add property to control packet duration
+ Add a property to specify that the amount of data in a packet should be a
+ multiple of ptime-multiple.
+ See #613248
- * po/LINGUAS:
- * po/de.po:
- Add German translation (#352069).
+2010-04-09 11:20:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-01 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 218568f to fc85867
- reviewed by: Wim Taymans <wim@fluendo.com>
+2010-04-08 17:49:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
- Use newly added GstCollectPads API to free the allocated resources in
- the GstOggPad structures (#402393).
+ * ext/ogg/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.h:
+ playback, ogg: dist new gstplayback.h and gstogg.h
-2007-01-31 Jan Schmidt <thaytan@mad.scientist.com>
+2010-04-09 08:23:33 +0200 Thomas Green <thomasgr33n@googlemail.com>
- * gst/playback/gstplaybin.c: (gen_vis_element):
- Add audioresample+audioconvert in front of the visualisation
- element, so that elements like libvisual 0.4 that don't support all
- samplerates can work.
+ * gst/playback/gstplaybin.c:
+ playbin: Only unref the volume element on dispose and when a new audio sink is set
+ Unreffing it whenever the sinks are removed will make the volume
+ element unavailable after a playbin reuse because it is only
+ recreated if the audio sink has changed.
+ Fixes bug #614288.
- Fixes: #402505
+2010-04-08 07:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-30 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Be sure that metadata is writable before setting caps
+ Call gst_buffer_make_metadata_writable before attempting
+ to set caps on the buffer.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
- (gst_play_base_bin_get_streaminfo_value_array):
- Take some locks and make a copy of the streaminfo value array we
- maintain while holding the lock, so that the application can
- retrieve the stream-info as a value array in a thread-safe way.
+2010-04-08 12:21:50 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-30 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgio.c:
+ * ext/gnomevfs/gstgnomevfs.c:
+ ext: Invert rank of gio and gnomevfs elements
- * gst/audioconvert/gstaudioconvert.c:
- Don't fail on 0 sized buffers. Fixes #396835.
+2010-04-08 01:26:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-29 David Schleef <ds@schleef.org>
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: don't pass non-constant strings as printf format strings
+ Fixes 'format not a string literal and no format arguments' compiler
+ warning when compiling with -DGST_DISABLE_PRINTF_EXTENSION.
- * gst/typefind/gsttypefindfunctions.c:
- Detect BBCD as video/x-dirac, so we can play raw dirac
- streams.
+2010-04-07 20:21:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/video/video.h:
+ docs: add gtk-doc chunks with Since: tags for new GST_VIDEO_CAPS_GRAY* API
- * ext/theora/theoraenc.c: (theora_enc_chain):
- Check return value of theora_encode_header(), or we might try to
- allocate a random number of bytes. theora_encode_header() can fail
- if libtheora has been compiled with encoding support disabled.
- Fixes #398110.
+2010-04-07 19:07:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-29 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ * configure.ac:
+ build: bump autoconf requirement to 2.60 for gobject-introspection.m4
+ Require autoconf 2.60 (which was released in June 2006).
+ Fixes #600718.
- * tests/check/gst/.cvsignore:
- Do as buildbot says.
+2010-04-07 17:25:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/video/video.c:
+ video: Fix parsing of 8-bit grayscale caps
- * ext/libvisual/visual.c: (gst_visual_src_setcaps):
- Fix strides in libvisual. Gst uses X strides.
- Inspired by: <ed at catmur dot co dot uk> and
- <tim at centricular dot net>
- Fixes #401118.
+2010-04-07 17:21:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-27 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.h:
+ video: API: Add GST_VIDEO_CAPS_GRAY{8,16}
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
- (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
- (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
- (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
- (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
- (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
- * ext/ogg/gstoggdemux.h:
- Properly propagate streaming errors when we are scanning the file for
- chains so that we don't crash when shut down. Might fix some crashers
- when quickly switching oggs in RB such as #332503 and #378436.
+2010-04-07 17:08:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-26 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ video: API: Add gst_video_format_is_gray() to the docs
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
- Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
- error code as well.
+2010-04-07 17:07:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-25 Wim Taymans <wim@fluendo.com>
+ * win32/common/libgstvideo.def:
+ video: Add new symbol to the exported symbols list
- * gst/playback/gstplaybasebin.c: (remove_source):
- Don't try to disconnect a signal from a finalized object.
+2010-04-07 17:06:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for 8-bit and 16-bit grayscale formats
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
- Cast lock macro parameters to make sure we're actually accessing the
- lock member at the right class level. Free list itself in _dispose()
- as well and NULL it in case dispose gets called multiple times.
+2010-04-06 10:55:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-25 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtspconnection: Handle closed POST socket in tunneling
+ Catch more socket errors.
+ Rework how sockets are managed in the GSource, wake up the maincontext instead
+ of adding/removing the sockets from the source.
+ Add callback for when the tunnel connection is lost. Some clients (Quicktime
+ Player) close the POST connection in tunneled mode and reopen the socket when
+ needed.
+ See #612915
- * gst/playback/gstdecodebin2.c:
- (gst_decode_bin_dispose),(gst_decode_bin_finalize):
- Free GstDecodeGroups no longer used.
- (gst_decode_group_expose):
- Don't unlock too many times !
- (deactivate_free_recursive):
- Free iterator once we're done with it.
- Fix for recursively deactivating elements (stop at ghostpads).
+2010-04-04 21:24:44 -0700 David Schleef <ds@schleef.org>
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure: fix cdparanoia check
+ Linking with libcdda_paranoia.so requires also linking with
+ libcdda_interface.so.
- * gst/playback/gstplaybin.c: (handoff):
- Fix up caps on the frame buffer before we save it and potentially
- make it accessible to other threads via g_object_get; also use
- gst_buffer_replace() instead of gst_mini_object_replace().
+2010-04-04 18:00:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/libs/tag.c:
+ tests: tag: Refactor a bit
+ Refactor xmp tags unit tests and remove an useless assertion.
+ This will make easier to add unit tests to serialize/deserialize
+ taglists.
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Make getting the current frame thread-safe.
+2010-04-04 21:18:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-25 Edward Hervey <edward@fluendo.com>
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: Ignore errors when unpreparing or closing the device
+ Errors could happen here when the device was removed already
+ or when something is broken anyway. If errors happen here and
+ they're propagated, the element can't shutdown cleanly.
+ Fixes bug #614545.
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
- (gst_decode_group_new), (gst_decode_group_free):
- Set queues to bigger sizes to cope with HD contents.
- Fix some mutex freeing and add comment about MT safe methods.
+2010-04-04 20:55:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-24 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: Detect errors from device polling, stop the task and post an error message
+ Partially fixes bug #614545.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
- (gst_text_overlay_text_event):
- Don't unnecessarily ref (and then leak) upstream events if the text
- pad is not linked. Fixes #399948.
+2010-04-04 12:13:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/gst-plugins-base.supp:
- Add suppression for pango on edgy/x86 for textoverlay test.
+ * configure.ac:
+ * tests/examples/seek/Makefile.am:
+ examples: build silly joystick seek example only on linux
+ jsseek depends on linux headers and should therefore only be built
+ on linux.
+ Fixes #614764.
-2007-01-24 Wim Taymans <wim@fluendo.com>
+2010-04-03 22:49:11 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Add some more fixed payloads.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: swap timestamps in forward and reverse mode.
+ In reverse mode we want use the next next timestamp (and not the other way
+ around). Fixes the tests again. Also readd a log line that was dropped with
+ previous commit.
-2007-01-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-03 14:03:45 +0100 Vincent Untz <vuntz@gnome.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
- Error out properly if we get an error from libogg while reading the
- BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ libs: point gobject-introspection scanner to .la files
+ Point g-ir-scanner to the .la file of our library, which hopefully
+ makes it find the right dependencies in all cases (ie. our locally
+ built libgstreamer and not the system-installed one). This is also
+ how it's done in Gtk+ and how it's documented in the wiki, see
+ http://live.gnome.org/GObjectIntrospection/AutotoolsIntegration
+ Fixes #603710.
-2007-01-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-02 21:01:25 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
- Don't leak mutex.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: implement reverse playback
+ Support playback at negative rates. When having a GstController assigned, the
+ element will produce time dependend output.
- * tests/check/elements/playbin.c:
- (test_sink_usage_video_only_stream),
- (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
- (test_suburi_error_wrongproto), (test_missing_urisource_handler),
- (test_missing_suburisource_handler),
- (test_missing_primary_decoder), (playbin_suite):
- Run all tests once with decodebin and once with decodebin2.
- One test does not pass yet with decodebin2.
+2010-04-02 20:56:19 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-01-23 Edward Hervey <edward@fluendo.com>
+ * tests/icles/audio-trickplay.c:
+ tests: extend audio-trickplay test app
+ Tell status in top comment. Use debug logging instead of print to be able to
+ see timing issue in debug log viewer. Add more commandline flags. Test reverse
+ playback.
- * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
- Fix the cases where oggmux doesn't properly figure out that all
- sinkpads have gone EOS, and therefore doesn't push out the remaining
- buffers and the final EOS event.
- Fixes #363379
+2010-04-02 18:56:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-23 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c:
+ seek: Only use embed_xid if HAVE_X is defined
+ Fixes bug #614622.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Don't lock on navigation event push, just on keysym to string.
- Fixes #397673 again.
+2010-04-01 19:13:22 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Edward Hervey <edward@fluendo.com>
+ * tests/check/pipelines/basetime.c:
+ tests/basetime: Don't run test with osxaudiosrc
+ libcheck runs the actual tests in a forked process and that makes the guys
+ in Cupertino really sad.
- * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
- (get_current_group), (group_demuxer_event_probe),
- (gst_decode_group_expose), (deactivate_free_recursive),
- (gst_decode_group_free):
- Cleanups.
- Don't forget to emit 'no-more-pads' once a group is exposed.
- Cleanup elements from a DecodeGroup once we remove it.
- Protect call to gst_decode_group_expose() with the decodebin lock.
+2010-04-01 18:51:17 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ tests: Unref the bus once we're done with it
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Looking at Xorg code i can't figure out if that XKeysymToString
- function is thread sensible or not. Lock it just in case as
- recommended by Radek Doulik <rodo at ximian dot com>.
+2010-04-01 16:49:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+ * common:
+ common: Update for new suppressions
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Lock that X Call as well. Fixes #397673.
+2010-04-01 13:55:15 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ gstplaysink: Remove unused variable.
+ The value of klass is never used
- * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
- Don't go into an endless loop if the file starts with 00 00 01 2X,
- like quicktime redirect files might. Fixes #396042.
+2010-04-01 13:53:37 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/Makefile.am:
- * tests/check/gst/.cvsignore:
- * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
- (typefindfunctions_suite):
- Add unit test for the above.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Removing dead assignment.
+ The value of group is overwritten a few lines below before being used.
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-01 13:51:13 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- On second thought, use "depth" field rather than "bpp" field.
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Remove unused variable
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2010-04-01 13:48:42 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Camtasia caps apparently need a bpp field (#398875).
+ * ext/gnomevfs/gstgnomevfssink.c:
+ gstgnomevfssink: Return the proper GstFlowReturn.
+ We were always returning GST_FLOW_OK previously even if we encountered errors.
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-30 23:44:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (setup_subtitle),
- (gen_source_element), (gst_play_base_bin_change_state):
- Attempt at a better error message in case we don't have the required
- URI handler installed; post missing-plugin message also when we're
- missing an URI handler for the subtitle URI; clean up properly also
- when an error occurs and we never made it to PAUSED state.
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ gst-libs: more gobject-introspection fixes
+ Use right .pc file variable for compiler includes this time:
+ g-ir-compiler wants the girdirs not the typelibdirs as includes.
- * tests/check/elements/playbin.c: (GST_START_TEST),
- (playbin_suite):
- Check that we're also getting a missing-plugin messsage for a
- missing subtitle URI handler (and clean up properly).
+2010-03-30 20:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/jsseek.c:
+ examples: fix printf format warning in jsseek example
+ Yes, I know about G_GSIZE_FORMAT.
- * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
- Plug a few reference leaks.
+2010-03-30 19:56:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ gst-libs: fix up gobject-introspection some more
+ Use new girdir and typlibdir from core .pc files, so we can figure
+ out the right includes to pass to the gobject-introspection tools,
+ whether core is installed in the same prefix as gobject-introspection
+ or in a different prefix or uninstalled. This also keeps us from adding
+ bogus paths to the includes that only work if core is uninstalled.
+ Also add some missing includes/pkgs where needed.
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Lower probability a bit if the marker isn't right at the start,
- to decrease the chance of false positives.
+2010-03-30 19:29:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/Makefile.am:
+ Our RIFF library depends on both the audio and tag libraries
+ Update rules in Makefile.am accordingly.
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Small mpeg2 system stream typefinding improvement: make typefinder
- probe a bit into the stream instead of just looking for a marker
- at the beginning. Fixes #397810.
+2010-03-30 15:10:42 +0200 Robert Swain <robert.swain@collabora.co.uk>
-2007-01-18 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Fix aduio_raw_sink typo
- * gst/audioconvert/gstchannelmix.c:
- Remove compatibility cruft for prehistoric GLib versions.
+2009-11-28 21:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-17 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/.gitignore:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/jsseek.c:
+ examples: Add a silly joystick based shuttle example
- * gst/playback/Makefile.am:
- * gst/playback/gstdecodebin.c: (close_pad_link):
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_handle_message_func), (unknown_type):
- Let decodebin be the element to post missing-plugin messages for
- missing decoders (rather than playbin); make playbin implement
- GstBin::handle_message so we can suppress missing-plugin messages
- for types we're not handling on purpose (don't want to bring up an
- installer in those cases).
+2010-03-29 20:07:52 -0700 David Schleef <ds@schleef.org>
-2007-01-16 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: 0-length packets are delta units
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_to_vorbiscomment_buffer):
- * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
- Fix potentially unaligned access (#397207).
+2010-03-29 10:47:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/Makefile.am:
+ gst-libs: build independent sub-directories in parallel if make -jN is used
+ Build those libraries that don't depend on any other gst-plugins-base
+ libraries in parallel if make -jN is used.
- * tests/examples/seek/seek.c: (set_scale), (update_scale),
- (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
- (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
- (main):
- Allow to toggle looping while it plays. Fix callback prototype. Clean
- up code a bit more. Add copyright header.
+2010-03-29 00:22:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ * ext/Makefile.am:
+ * gst/Makefile.am:
+ * sys/Makefile.am:
+ * tests/examples/Makefile.am:
+ build: build plugin and example directories in parallel if make -jN is used
+ We know our plugins and examples are independent of each other, so may
+ just as well build them in parallel. Makes the output a bit messy, but
+ that shouldn't be a problem and can easily be avoided with make -j1.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Red and blue mask was swapped (spotted by Dan Williams).
+2010-03-28 21:50:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-15 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/Makefile.am:
+ gst-libs: specify dependencies in Makefile.am to make them explicit
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Use new beats-per-minute tag from core.
+2010-03-24 09:59:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ * sys/xvimage/xvimagesink.c:
+ * tests/icles/test-xoverlay.c:
+ xoverlay: change new set_render_rectangle() vfunc to take four arguments so we don't depend on libgstvideo
+ Don't make libgstinterfaces (and thus libgstaudio etc.) indirectly depend
+ on libgstvideo by using the GstVideoRectangle helper structure in the API,
+ which causes undesirable dependencies, esp. with the gobject-introspection
+ (people will point and laugh at us if they find out that libgstaudio
+ depends on libgstvideo). Instead, pass the x, y, width and height parameters
+ directly to the function.
+ Re-fixes #610249.
+
+2010-03-25 18:45:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * po/POTFILES.in:
- Add new files with translatable strings, so they actually make it
- into the template file one day.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: we can handle avi in download mode too
+ Add avi to the whitelisted types that can be used for download buffering.
-2007-01-12 Andy Wingo <wingo@pobox.com>
+2010-03-26 15:57:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
- (gst_base_audio_sink_activate_pull): Remove the handwavey nego
- stuff, as the base class handles this now. Actually tell the ring
- buffer to start.
- (gst_base_audio_sink_callback): Cast the ring buffer correctly.
- How did this work before? Maybe I'm not as awesome a programmer as
- I think.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Provide packet duration function for old FLAC mapping too
+ Fixes bug #613809.
- * gst-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
- of a pad function.
+2010-03-18 22:12:40 +0000 Damien Lespiau <damien.lespiau@intel.com>
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ * autogen.sh:
+ autogen.sh: Don't call configure with --enable-plugin-docs
+ configure gives a nice warning:
+ configure: WARNING: unrecognized options: --enable-plugin-docs
+ and indeed, I could not find anything in the configure.ac or the m4
+ macros that would allow enabling that option. Remove it then.
- * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
- Remove more fields so that the application can better blacklist
- formats that have been tried before.
+2010-03-24 23:04:43 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Do not remove tag from list twice
+ There was a but when parsing the tags that removed two tags
+ from the list when only one was parsed
- * gst-libs/gst/audio/mixerutils.h:
- Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
- used when compiling with c++ compilers as well.
+2010-03-24 14:43:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Add some comments
+ Just adds some comments explaining some stuff about the
+ (de)serialization functions. Add myself to the copyright list too.
- * gst/typefind/gsttypefindfunctions.c:
- Fix comment.
+2010-03-24 10:18:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds _USER_RATING mapping for xmp
+ Adds a new mapping for _USER_RATING on xmp helper lib
+ and also adds tests for it
- * gst/playback/gstplaybin.c: (post_missing_element_message),
- (gen_video_element), (gen_text_element), (gen_audio_element),
- (gen_vis_element):
- Post missing-plugin messages also when we error out because
- converters, textoverlay or auto*sinks are missing (#161922).
+2010-03-23 09:32:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-01-10 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Add Elevation tag mapping
+ Adds a mapping to the _ELEVATION tag, this is a different
+ mapping as it has to be mapped into exif:GPSAltitude and
+ exif:GPSAltitudeRef at the same time. So we needed to refactor
+ a little more to be able to deserialize it properly.
+ Now, when parsing a xmp buffer into a taglist all tags are
+ added to a list before being parsed so that when one of the
+ altitude tags are found the deserialization function can search
+ for its complementary tag to do the correct parsing
+ Fixes #613690
+
+2010-03-23 09:48:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Fix off by one
+ Avoid ignoring single char tags, like exif:GPSAltitudeRef
+ Fixes #613690
+
+2010-03-22 15:18:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ * tests/check/libs/tag.c:
+ tag: xmp: Adds mappings for LATITUDE and LONGITUDE
+ Adds the mappings for those tags and tests
+ for tags serialization.
+ Fixes #613690
+
+2010-03-22 22:03:09 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Refactor buffer parsing
+ When parsing the xmp buffer into the gst taglist store the
+ found tags into a list to be parsed only after finding all
+ tags on the buffer. This allows the parser function to search
+ this list for complimentary tags that should be parsed together
+ Fixes #613690
+
+2010-03-20 11:17:38 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ tag: xmp: Refactor mappings storage
+ This commit is only refactoring, no fetaures added.
+ Do not store tags in flexible arrays as it doesn't allow us
+ to use nested flexible arrays. This is going to be needed in the
+ following commits to map gst tags that are stored into
+ 2 separate tags in xmp (Not that they are alternatives, but
+ they are complementary).
+ For example, GST_TAG_ELEVATION is represented in the exif
+ schema with 2 fields: the absolute altitude and an integer
+ to indicate if it is above or below sea level.
+ The previous mappings storage wouldn't allow us to
+ express it.
+ Also store a serialization and a deserialization function
+ for each xmp tag as some of them require some non-trivial
+ convertion to its string form.
+ Fixes #613690
+
+2010-03-24 18:51:42 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 55cd514 to c1d07dd
+
+2010-03-24 18:55:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
- (is_demuxer_element), (new_caps):
- * gst/playback/gstplaybasebin.c: (source_new_pad):
- Fix the case where we try to ref a NULL element when we delay a link
- because of unfixed caps.
- Set the state of autoplugged decodebins to PAUSED.
- RTSP now works in playbin, we can remove it from the blacklist.
+ * tests/examples/seek/seek.c:
+ seek: parse more info from the buffering query
+ Parse more info from the buffering query and log this as debug info.
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-24 12:10:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstplaybasebin.c: (string_arr_has_str),
- (unknown_type), (setup_subtitle), (gen_source_element):
- * gst/playback/gstplaybin.c: (plugin_init):
- Post missing-plugin messages on the bus for missing sources and
- missing decoders/demuxers/depayloaders; fix error code used when
- we're missing an URI handler source; for media types that we are not
- handling on purpose at the moment, don't print "don't know how to
- handle xyz" messages to the terminal or post missing-plugin
- messages on the bus.
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsptransport: ignore unparsable ranges
+ Ignore unparsable port ranges instead of erroring out.
+ Fixes #613591
- * tests/check/elements/playbin.c: (create_playbin),
- (GST_START_TEST), (gst_codec_src_uri_get_type),
- (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
- (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
- (gst_codec_src_init_type), (gst_codec_src_base_init),
- (gst_codec_src_create), (gst_codec_src_class_init),
- (gst_codec_src_init), (plugin_init), (playbin_suite):
- Add some tests for the missing-plugin stuff.
+2010-03-23 18:36:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * win32/common/libgstrtsp.def:
+ win32: Add new gst_rtsp_lower_trans_get_type() symbol to the symbol lists
+
+2010-03-23 11:01:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add some more fourcc for MPEG-4 video
+
+2010-03-22 09:15:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/utils/Makefile.am:
- * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
- * gst-libs/gst/utils/base-utils.h:
- * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
- (find_format_info), (caps_are_rtp_caps),
- (gst_base_utils_get_source_description),
- (gst_base_utils_get_sink_description),
- (gst_base_utils_get_decoder_description),
- (gst_base_utils_get_encoder_description),
- (gst_base_utils_get_element_description),
- (gst_base_utils_add_codec_description_to_tag_list),
- (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
- * gst-libs/gst/utils/descriptions.h:
- * gst-libs/gst/utils/missing-plugins.c:
- (missing_structure_get_type), (copy_and_clean_caps),
- (gst_missing_uri_source_message_new),
- (gst_missing_uri_sink_message_new),
- (gst_missing_element_message_new),
- (gst_missing_decoder_message_new),
- (gst_missing_encoder_message_new),
- (missing_structure_get_string_detail),
- (missing_structure_get_caps_detail),
- (gst_missing_plugin_message_get_installer_detail),
- (gst_missing_plugin_message_get_description),
- (gst_is_missing_plugin_message):
- * gst-libs/gst/utils/missing-plugins.h:
- API: add new libgstbaseutils library with functions
- - to create and parse missing-plugins messages
- - that provide (translated) descriptions for caps/decoders/sources/etc.
- Closes #392393.
+ configure: require core git
+
+2010-03-22 08:38:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft.pc.in:
+ pkgconfig: Add @LIBM@ to the FFT pkg-config files
+
+2010-03-22 08:35:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-audio-uninstalled.pc.in:
+ * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast.pc.in:
+ * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
+ * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
+ * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Add new lib.
+ * pkgconfig/gstreamer-riff-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-tag-uninstalled.pc.in:
+ * pkgconfig/gstreamer-video-uninstalled.pc.in:
+ pkgconfig: Fix include and library paths for the uninstalled pc files
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Generate docs for new lib and API.
+2010-03-20 13:42:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/libs/.cvsignore:
- * tests/check/libs/utils.c: (missing_msg_check_getters),
- (GST_START_TEST), (libgstbaseutils_suite):
- Add some basic unit tests.
+ * ext/gio/gstgiobasesrc.c:
+ gio: add cast to avoid compiler warning with old GLib versions
+ g_file_input_stream_query_info() had char * instead of const char *
+ as attribute argument before 2.20.
+ Fixes #613387, spotted by tetsuyayasuda@gmail.com
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-20 12:55:36 +0000 Torsten Schönfeld <kaffeetisch@gmx.de>
- * ext/ogg/Makefile.am:
- Dist gstoggdemux.h to fix 'make distcheck'.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ docs: add Since: tags to gst_x_overlay_handle_event() docs
+ Fixes #613403.
- * sys/v4l/Makefile.am:
- Fix 'make distcheck' even more.
+2010-03-19 22:33:58 +0100 Benjamin Otte <otte@redhat.com>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Constify some strings in the API
+ Needed by plugins-good
-2007-01-09 Wim Taymans <wim@fluendo.com>
+2010-03-19 16:41:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
- (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
- (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
- (gst_ogg_demux_perform_seek):
- * ext/ogg/gstoggdemux.h:
- Added docs.
- Add some more comments.
- Small cleanups.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Only set color-matrix and chroma-site for relevant formats
+ The color-matrix only makes sense for colorful formats, i.e. not Y800
+ and the chroma-site only for non-4:4:4(:4) formats.
-2007-01-09 Wim Taymans <wim@fluendo.com>
+2010-03-19 15:37:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/theoradec.c:
- * ext/vorbis/vorbisdec.c:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full):
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Small documentation updates/fixes
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoradec.h:
+ theoradec: add QoS messages to the decoder
+ Post QoS messages when we drop a frame because of QoS.
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-19 15:00:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Require core CVS HEAD for Andy's basesrc/sink API additions.
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: add GType for transport flags
+ Make a method to register the transport flags as a GType.
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-19 01:00:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Günter Thelen <daedalus dot inc at gmx net>
+ * ext/cdparanoia/Makefile.am:
+ * ext/gio/Makefile.am:
+ * ext/gnomevfs/Makefile.am:
+ * ext/libvisual/Makefile.am:
+ * ext/ogg/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ * gst/ffmpegcolorspace/Makefile.am:
+ * gst/tcp/Makefile.am:
+ * gst/videotestsrc/Makefile.am:
+ * sys/v4l/Makefile.am:
+ * tests/examples/app/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tests/icles/Makefile.am:
+ build: Makefile.am fixes
+ Mostly just add missing $(GST_BASE_CFLAGS), but also fix up order
+ of flags (see docs/random/moving-plugins).
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
- (plugin_init):
- Add typefinder for flac-in-ogg in conformance with the ogg-mapping
- on flac.sf.net (there appear to be other versions of the first
- ogg page in the wild) (#391365).
+2010-03-19 00:46:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore new unit test binary
+
+2010-03-17 23:57:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- Check if localtime_r() is available.
+ configure.ac: -Wmissing-prototypes and -Wnested-externs are not valid for C++
+ Fixes building Qt-based overlay examples in combination with -Werror.
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
- If localtime_r() is not available, fall back to localtime(). Should
- fix build on MingW (#393310).
+2010-03-17 16:32:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure.ac: wrap overly long warning flag lines
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * gst/subparse/gstsubparse.h:
- Remove spurious 1000 subtrahend when calculating the timestamp from
- the frame number and the frame rate . Also, use the frames/second
- value specified in the first line of the file, if one is specified
- there. Should fix #357503.
+2010-03-17 19:24:27 -0300 Reuben Dowle <reube.dowle@navico.com>
- * tests/check/elements/subparse.c: (do_test),
- (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
- (subparse_suite):
- Add some basic unit tests for the microdvd subtitle format.
+ * sys/ximage/ximagesink.c:
+ ximagesink: Fix caps leak
+ Unref caps when peer doesn't accept caps
+ Fixes #613198
-2007-01-07 Julien MOUTTE <julien@moutte.net>
+2010-03-17 08:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ tests: capsfilter-renegotiation: Adds a new unit test
+ Adds a new test for checking that capsfilter 'caps' property
+ changes cause caps renegotiation on the pipeline.
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xvimage_put), (gst_lookup_xv_port_from_adaptor),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_set_event_handling),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- Fixes: #390076.
- Add an adaptor property to select a specific XV adaptor.
- * sys/xvimage/xvimagesink.h:
+2010-03-17 16:46:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-07 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
- (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
- (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
- (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_change_state), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
- Use flow_lock much more to protect every access to xwindow.
- Try to catch erros while creating images in case some drivers are
- just generating an XError when the requested image is too big.
- Should fix : #354698, #384008, #384060.
- * tests/icles/stress-xoverlay.c: (cycle_window), (create_window):
- Implement some stress testing of setting window xid.
-
-2007-01-07 Sébastien Moutte <sebastien@moutte.net>
-
- * win32/common/libgsaudio.def:
- Add new exported function.
- * win32/common/libgstogg.dsp:
- Add gstoggaviparse.c to the build.
- * win32/common/libgstvideoscale.dsp:
- Add vs_4tap.c to the build.
- * win32/common/libgstvorbis.dsp:
- Add vorbistag.c to the build.
-
-2007-01-06 Andy Wingo <wingo@pobox.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_class_init)
- (gst_base_audio_sink_init):
- (gst_base_audio_sink_activate_pull): Add an activate_pull function
- to baseaudiosink, and tell basesink that we can work in pull mode.
- This way the ring buffer thread drives the pipeline directly, if
- pull mode is possible. There is some lingering nastiness regarding
- capsnego, however.
- (gst_base_audio_sink_callback): Implement the callback to pull
- data. This interface is a bit light, though -- it should get a
- GstFlowReturn return value at least.
-
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * gst/playback/gstdecodebin2.c:
- (gst_decode_group_check_if_blocked):
- Printf format and missing argument fixes.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Use correct boundary checks for YUY2/UYVY
+ Fixes bug #613093.
-2007-01-05 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-17 16:39:13 +0100 Peter Kjellerstedt <peter.kjellerstedt@axis.com>
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
- (gst_ogm_parse_change_state):
- Activate pads before adding them to the element.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Further clean up of gst_rtsp_strresult()
+ Since we no longer use an array of error messages, there is no reason
+ to clamp the error code, which allows us to simplify the code some more
+ and also to actually report the correct error code for unknown errors.
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 15:41:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/scrubby.c: (main):
- * tests/examples/seek/seek.c: (main):
- Call g_thread_init() first thing in main() (see #391278).
+ * gst/volume/gstvolume.c:
+ volume: Remove useless cast
+ It's not necessary anymore after latest core change to GstValueArray.
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 12:08:30 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/Makefile.am:
- * tests/check/libs/.cvsignore:
- * tests/check/libs/netbuffer.c: (GST_START_TEST),
- (netbuffer_suite):
- Add test for GstNetBuffer + gst_buffer_copy(). Disabled
- for the time being, since it's broken, see #393099.
+ * configure.ac:
+ Add more warning flags
+ The warnings are:
+ -Wcast-align
+ -Winit-self
+ -Wmissing-include-dirs
+ -Waddress
+ -Waggregate-return
+ -Wno-multichar
+ -Wnested-externs
+ No code needed to be fixed.
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 11:14:29 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/Makefile.am:
- Update to use GST_PLUGINS_BASE_CFLAGS as well.
+ * gst/audioconvert/gstfastrandom.h:
+ Fix for -Wold-style-definition
+ I didn't add the flag to configure because libvisual ships headers that
+ trigger this warning.
-2007-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-03-17 10:53:21 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
- split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
- so that GST_BASE_CFLAGS can go inbetween them, making sure
- we use uninstalled gst-libs headers
- * docs/libs/Makefile.am:
- * ext/alsa/Makefile.am:
- * ext/cdparanoia/Makefile.am:
- * ext/gnomevfs/Makefile.am:
- * ext/libvisual/Makefile.am:
- * ext/ogg/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst/adder/Makefile.am:
- * gst/audioconvert/Makefile.am:
- * gst/audiorate/Makefile.am:
- * gst/audioresample/Makefile.am:
- * gst/playback/Makefile.am:
- * gst/tcp/Makefile.am:
- * gst/videoscale/Makefile.am:
- * gst/volume/Makefile.am:
- * sys/ximage/Makefile.am:
- * sys/xvimage/Makefile.am:
- * tests/icles/Makefile.am:
- adapt
-
-2007-01-04 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/interfaces/xoverlay.c:
- (gst_x_overlay_handle_events):
- * gst-libs/gst/interfaces/xoverlay.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_set_event_handling),
- (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
- (gst_ximagesink_get_property), (gst_ximagesink_init),
- (gst_ximagesink_class_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_set_event_handling),
- (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
- (gst_xvimagesink_get_property), (gst_xvimagesink_init),
- (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- * tests/icles/stress-xoverlay.c: (toggle_events), (create_window):
- Add a method to the XOverlay interface to allow disabling of
- event handling in x[v]imagesink elements. This will let X events
- propagate to parent windows which can be usefull in some cases.
- Be carefull that the application is then responsible of pushing
- navigation events and expose events to the video sink.
- Fixes: #387138.
+ * ext/pango/gstclockoverlay.h:
+ * gst/subparse/mpl2parse.c:
+ Add -Wformat-nonliteral -Wformat-security
+ And fix the resulting compile failures.
+ I'm sorry about the patch necessary to gstclockoverlay.h but after
+ talking to Tim we decided we can live with it.
-2007-01-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 10:51:57 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
- (fixes #392070).
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Refactor gst_rtsp_strresult
+ 2 goals in the refactoring:
+ - Put the error messages closer to their enum values, so that it's easy
+ to see which error belongs to which value.
+ - Make gcc not complain with -Wformat-nonliteral
-2007-01-01 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 10:47:07 +0100 Benjamin Otte <otte@redhat.com>
- * configure.ac:
- * docs/Makefile.am:
- * docs/design/Makefile.am:
- Dist design docs.
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmp: Refactor code
+ I initially looked here because I wanted compiles to not fail with
+ -Wformat-nonliteral but ended up refactoring the code to make it look
+ nicer.
+ As I lack a large collection of XMP tagged files, I only did rough
+ testing of the code. The testsuite passes though.
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+2010-03-16 20:05:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation
- typo. Fixes: #390063.
+ * m4/Makefile.am:
+ * m4/a52.m4:
+ * m4/aalib.m4:
+ * m4/as-arts.m4:
+ * m4/as-ffmpeg.m4:
+ * m4/as-liblame.m4:
+ * m4/as-slurp-ffmpeg.m4:
+ * m4/esd.m4:
+ * m4/gconf-2.m4:
+ * m4/glib.m4:
+ * m4/gst-artsc.m4:
+ * m4/gst-matroska.m4:
+ * m4/gst-sdl.m4:
+ * m4/gst-shout2.m4:
+ * m4/gst-sid.m4:
+ * m4/gtk.m4:
+ * m4/libfame.m4:
+ * m4/libmikmod.m4:
+ m4: remove some unused .m4 files
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+2010-03-16 18:31:15 +0100 Benjamin Otte <otte@redhat.com>
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
- caps leak.
- * win32/common/config.h: Updated.
+ * ext/alsa/gstalsaplugin.c:
+ * ext/ogg/gstoggdemux.c:
+ More ENABLE_NLS fixes
-2006-12-22 Stefan Kost <ensonic@users.sf.net>
+2010-03-16 18:06:16 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
- (setup_gdpdepay_streamheader):
- * tests/check/elements/gdppay.c: (cleanup_gdppay),
- (setup_gdppay_streamheader):
- Fix the dp tests, but activating the pads for the streamheader tests
- too and cleaning up conditionaly
+ * gst-libs/gst/gettext.h:
+ Fix for ENABLE_NLS being undefined for -Wundef
-2006-12-22 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-15 22:49:53 +0100 Benjamin Otte <otte@redhat.com>
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
- (img_get_alpha_info):
- Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
- other end of the word. Fixes: #387073.
+ * configure.ac:
+ * ext/libvisual/visual.c:
+ * ext/theora/gsttheoraenc.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ * gst-libs/gst/tag/tags.c:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ * gst/volume/gstvolume.c:
+ * sys/v4l/gstv4lelement.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/gdpdepay.c:
+ * tests/check/elements/playbin.c:
+ * tests/check/elements/playbin2.c:
+ * tests/check/elements/videorate.c:
+ * tests/check/libs/pbutils.c:
+ * tests/check/libs/video.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ * tests/icles/stress-playbin.c:
+ Add -Wwrite-strings to configure
+ Fixes for the code included
- Add some inconsequential branch hints in a couple of places.
+2010-03-16 15:45:23 +0100 Benjamin Otte <otte@redhat.com>
-2006-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsamixerelement.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/libvisual/visual.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisdeclib.h:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gststreamselector.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * sys/v4l/gstv4ljpegsrc.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * sys/v4l/gstv4lsrc.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/playbin.c:
+ * tests/check/elements/playbin2.c:
+ * tests/check/elements/textoverlay.c:
+ * tests/check/libs/cddabasesrc.c:
+ * tests/check/libs/pbutils.c:
+ * tests/old/testsuite/alsa/formats.c:
+ * tests/old/testsuite/alsa/sinesrc.c:
+ gst_element_class_set_details => gst_element_class_set_details_simple
+ Also change my email from the old university one to the current one.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_smpfmt):
- The "signed" field in raw audio caps is of boolean type, trying to
- extract the value with _get_int() will fail (fix to keep in sync with
- the copy in gst-ffmpeg)
+2010-03-15 22:17:56 +0100 Benjamin Otte <otte@redhat.com>
-2006-12-21 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/elements/audioresample.c: (cleanup_audioresample):
- * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
- * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
- (cleanup_gdpdepay):
- * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
- * tests/check/elements/subparse.c: (teardown_subparse):
- * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
- * tests/check/elements/videorate.c: (cleanup_videorate):
- * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
- * tests/check/elements/volume.c: (cleanup_volume):
- * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
- (cleanup_vorbisdec):
- * tests/check/elements/vorbistag.c: (setup_vorbistag),
- (cleanup_vorbistag):
- consistent pad (de)activation
+ * configure.ac:
+ Add -Wundef flag
-2006-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-16 16:15:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Forgot to register the extensions.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: allow for more ipv6 addresses
+ Use hints in getaddrinfo() so that we can also resolve ipv6 addresses.
-2006-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-11 14:52:09 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
- (plugin_init):
- Add typefinder for VIVO files (my christmas present to the 90s).
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
+ Fixes #612223.
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-16 01:08:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (type_found):
- Special-case the text/plain media type: we only want to recognise it
- as a 'raw' decoded media type if it comes from a demuxer or subtitle
- parser, but not if the entire stream is of text/plain type. If the
- entire stream is text/plain, we should just error out.
+ * tests/check/elements/videorate.c:
+ tests: fix videorate test
+ Fix up videorate test for latest videotestsrc changes: just check for
+ the important bits in the negotiated caps, not for exact equality with
+ our filter caps. Also don't leak the videorate element in the test.
- This fixes playback of audio files with lyrics in totem. Totem can't
- distinguish between text files and subtitle files and passes any
- .txt file with the same basename as the main file to playbin as
- suburi, and playbin will then throw a 'subtitle found, but no video
- stream' error, which isn't entirely helpful. See #380342.
+2010-03-15 12:54:32 -0500 Rob Clark <rob@ti.com>
- Also, with this change we'll show a slightly more correct error
- message in case totem passes a playlist file to us (although a
- custom error message wording instead of the default text would
- probably not be a bad idea either).
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP7 fourccs
+ Fixes #612968.
- Same problem also needs to be fixed for playbin+decodebin2.
+2010-03-15 12:54:01 -0500 Rob Clark <rob@ti.com>
- * tests/check/Makefile.am:
- * tests/check/elements/decodebin.c: (src_handoff_cb),
- (decodebin_new_decoded_pad_cb), (GST_START_TEST),
- (decodebin_suite):
- Add simple unit test for decodebin for the above.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP62 fourcc
+ See #612968.
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 23:46:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
- Refuse to change state to READY when we failed to create any of the
- required elements in our instance init function.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/multichannel.c:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/interfaces/tuner.c:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ docs: more helper libraries docs fixes
+ Quieten gtk-doc a bit more.
-2006-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 23:47:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
- Small docs fixes/updates.
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ docs: add GstRTSPExtension to docs
+ Add minimal docs for GstRTSPExtension so people know it exists.
- * gst-libs/gst/video/gstvideosink.h:
- Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
- from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
- removed from the base sink API between 0.9.6 and 0.9.7).
- API: add GST_VIDEO_SINK_CAST and use it for the height/width
- accessor macros, so we don't do a runtime GObject type check every
- time we use them.
+2010-03-15 18:45:13 +0000 David Hoyt <dhoyt@llnl.gov>
-2006-12-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: use g_ascii_strncasecmp() instead of strncasecmp()
+ g_ascii_strncasecmp() is more portable and likely more robust as
+ well (with random binary data as input).
+ Fixes #612845.
- * Makefile.am:
- * gst-plugins-base.doap:
- * gst-plugins-base.spec.in:
- add doap file
+2010-03-15 13:39:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-12-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstxmptag.c:
+ docs: fix typo in gst_tag_list_from_xmp_buffer() docs chunk
- Patch by: Jens Granseuer <jensgr at gmx net>
+2010-03-15 13:32:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
- Declare variables at the beginning of a block. Fixes #383195.
+ * gst-libs/gst/interfaces/navigation.c:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ docs: fix up interfaces library docs to make gtk-doc happy
-2006-12-07 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-15 13:24:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Bump version nano - back to CVS.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: add new libgstvideo API to documentation
+2010-03-15 13:19:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.10.11 ===
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstvideo.def:
+ win32: add recently added API to .def files
+ Also add API markers to make life easier for the release manager:
+ API: gst_x_overlay_set_render_rectangle()
+ API: gst_video_parse_caps_color_matrix()
+ API: gst_video_parse_caps_chroma_site()
-2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-15 13:14:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- releasing 0.10.11, "Dumb things"
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: use C comments instead of C++-style comments
-2006-12-05 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-15 13:10:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
- (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
- Handle the case where an element has multiple pads with
- unfixed caps as well as still possibly producing more dynamic
- pads by storing each case as a distinct entry in the dynamic list.
- Fixes #38223 again.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: use g_value_set_static_string() for string constants
-2006-12-04 Wim Taymans <wim@fluendo.com>
+2010-03-15 14:26:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix #382223, add more dynamic caps handling.
+ * gst/playback/gstplaysink.c:
+ playsink: Avoid g_object_set() on NULL if a text sink is used
+ Fixes bug #611702.
-2006-12-04 Wim Taymans <wim@fluendo.com>
+2010-03-15 14:10:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_init),
- (gst_netaddress_set_ip4_interface),
- (gst_netaddress_set_ip6_interface), (gst_netaddress_set_loopback),
- (gst_netaddress_set_ttl), (gst_netaddress_get_ip4_interface),
- (gst_netaddress_get_ip6_interface), (gst_netaddress_get_loopback),
- (gst_netaddress_get_ttl):
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- * gst/playback/gstdecodebin.c: (close_pad_link):
- * tests/examples/seek/seek.c: (end_scrub), (end_seek), (do_seek),
- (seek_cb), (stop_seek), (rate_spinbutton_changed_cb):
- * win32/common/config.h:
+ * gst/subparse/gstsubparse.c:
+ subparse: Correctly escape brackets in DKS regex
+ Fixes bug #612783.
-2006-12-01 Michael Smith <msmith@fluendo.com>
+2010-03-15 11:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Delete bad debug code.
- Fixes #381219
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: make timeout usec more accurate
+ Adjust the returned usec from the elapsed time so it represents the remaining
+ timeout.
-2006-12-01 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-15 11:41:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/videoscale/vs_4tap.c:
- * win32/MANIFEST:
- * win32/common/config.h:
- * win32/vs8/libgstvideoscale.vcproj:
- Fix compilation on win32 under VS8
- Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
- Partially fixes #381175
+ * tests/check/elements/videorate.c:
+ tests: update videorate test for videotestsrc changes
+ Add color-matrix to the caps we are comparing. Add logging og the caps in the
+ test.
-2006-11-30 Michael Smith <msmith@fluendo.com>
+2010-03-15 01:35:15 -0700 David Schleef <ds@schleef.org>
- * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
- (GST_START_TEST):
- It would be very bad if, after a discont buffer, we thought every
- single following buffer was also discont. So, add to the test to
- ensure that this isn't the case.
-
- * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
- ... it was the case. So fix it.
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add chroma-zone-plate pattern
+ pattern=chroma-zone-plate is pattern similar to zone-plate,
+ but in the chroma channels instead of luma.
-2006-11-28 Wim Taymans <wim@fluendo.com>
+2010-03-15 01:34:09 -0700 David Schleef <ds@schleef.org>
- * gst/playback/gstplaybasebin.c: (check_queue_event):
- Improve debug.
+ * ext/theora/gsttheoradec.c:
+ theoradec: add chroma-site to caps
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
- Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
- padtemplate caps. Refixes #357577.
+2010-03-15 01:33:36 -0700 David Schleef <ds@schleef.org>
-2006-11-28 Wim Taymans <wim@fluendo.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: add chroma-site to caps
- * gst/playback/gstplaybasebin.c: (check_queue_event),
- (queue_threshold_reached), (queue_out_of_data),
- (gen_preroll_element):
- Add event probe to see when EOS is in a queue and we can disable the
- underrun signals. Fixes #357577.
+2010-03-15 01:31:20 -0700 David Schleef <ds@schleef.org>
-2006-11-28 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: add gst_video_parse_caps_chroma_site()
- * gst/playback/Makefile.am:
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
- (_gst_boolean_accumulator), (gst_decode_bin_class_init),
- (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (gst_decode_bin_finalize), (gst_decode_bin_set_property),
- (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
- (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
- (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
- (connect_element), (expose_pad), (type_found),
- (pad_added_group_cb), (pad_removed_group_cb),
- (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
- (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
- (are_raw_caps), (multi_queue_overrun_cb),
- (multi_queue_underrun_cb), (gst_decode_group_new),
- (get_current_group), (group_demuxer_event_probe),
- (gst_decode_group_control_demuxer_pad),
- (gst_decode_group_control_source_pad),
- (gst_decode_group_check_if_blocked),
- (gst_decode_group_check_if_drained), (gst_decode_group_expose),
- (gst_decode_group_hide), (gst_decode_group_free),
- (gst_decode_group_set_complete), (source_pad_blocked_cb),
- (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
- (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
- (plugin_init):
- New decodebin2 element.
- Closes #370092
- * gst/playback/gstplay-marshal.list:
- Added marshallers for new signals in decodebin2
- * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
- Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
- is set.
+2010-03-14 19:10:16 -0700 David Schleef <ds@schleef.org>
-2006-11-28 Wim Taymans <wim@fluendo.com>
+ * ext/theora/gsttheoradec.c:
+ theoradec: add color-matrix to caps
- * gst/playback/gstplaybasebin.c: (setup_source),
- (gst_play_base_bin_change_state):
- Disable rtsp:// uris for the release, it's not good enough yet.
- Remove unused var.
+2010-03-14 16:17:46 -0700 David Schleef <ds@schleef.org>
-2006-11-26 Wim Taymans <wim@fluendo.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add color-matrix to template caps
- * ext/theora/theoradec.c: (gst_theora_dec_reset),
- (theora_dec_push_forward), (theora_dec_push_reverse),
- (theora_handle_data_packet), (theora_dec_decode_buffer),
- (theora_dec_flush_decode), (theora_dec_chain_reverse),
- (theora_dec_chain_forward), (theora_dec_chain):
- Implement reverse playback.
+2010-03-14 22:14:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
- (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
- (vorbis_dec_chain_forward):
- Clear buffers used for reverse playback in _reset.
- No need to set the eos flag, we clip samples using the segment.
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/examples/seek/seek.c:
+ * tests/icles/test-colorkey.c:
+ * tests/icles/test-xoverlay.c:
+ tests: make Gtk+ test programs compile with -DGSEAL_ENABLE
+ Fixes #612552, at least for now.
-2006-11-24 Wim Taymans <wim@fluendo.com>
+2010-03-14 22:13:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
- (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
- (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
- (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
- Some cleanups.
- Handle continued pages in reverse mode.
+ * Makefile.am:
+ build: add cruft alert for common/shave* leftovers to top-level Makefile.am
-2006-11-24 Wim Taymans <wim@fluendo.com>
+2010-03-14 13:11:53 -0700 David Schleef <ds@schleef.org>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
- (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
- (vorbis_dec_flush_decode):
- Small cleanups.
- Don't try to add invalid timestamps.
- Clipping will unref the buffer.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't drop zero-sized packets
+ Zero-sized packets have relevence to Theora.
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+2010-03-12 15:47:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.h:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- remove obsolete _factory_init protos
+ * gst/volume/gstvolume.c:
+ volume: Revert rounding behaviour changes when using controlled volume properties
+ Now the controlled and non-controlled code paths are all having
+ exactly the same rounding behaviour and the unit tests pass again.
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+2010-03-12 15:44:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
- Fix spacing in debug message.
+ * gst/volume/gstvolume.c:
+ volume: Only allocate a mute value array if a control source exists for the mute property
-2006-11-23 Wim Taymans <wim@fluendo.com>
+2010-03-12 13:55:55 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
- (gst_ogg_demux_chain):
- Don't just ignore return values from _pad_push().
- Small debug improvements.
+ * common:
+ Automatic update of common submodule
+ From e272f71 to 55cd514
-2006-11-23 Michael Smith <msmith@fluendo.com>
+2010-03-10 10:50:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
- If our incoming buffer is marked as DISCONT, then increment the page
- number (so that the discontinuity is marked in the final ogg
- bitstream) and flush the previous page.
+ * gst-libs/gst/tag/gstxmptag.c:
+ tags: Add new mapping to XMP helpers
+ Adds geotagging mappings to XMP helpers
+ Fixes #609539
-2006-11-22 Michael Smith <msmith@fluendo.com>
+2010-03-11 20:16:44 +0100 Benjamin Otte <otte@redhat.com>
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c: (gst_theora_enc_init),
- (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
- (theora_buffer_from_packet), (theora_enc_is_discontinuous),
- (theora_enc_chain), (theora_enc_change_state):
- Mark discontinuities of > 3/4 of a frame, reinit encoder.
+ * gst-libs/gst/interfaces/Makefile.am:
+ Don't have 2 include dirs
+ Seems to have been accidentally introduced in
+ 7269bc26d0a4bf44bd77a039fb54777625ef5f39.
- * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
- (GST_START_TEST), (theoraenc_suite):
- Enable discontinuity test, fix it.
+2010-03-11 16:35:10 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/audio-trickplay.c:
+ tests: Fix another unitialized variable
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
- (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
- (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
- (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
- (gst_text_overlay_change_state):
- * ext/pango/gsttextoverlay.h:
- Some textoverlay fixes: for one, in the video chain function,
- actually wait for a text buffer to come in if there is none at the
- moment and there should be one; also, deal more gracefully with
- incoming buffers that do not have a timestamp or duration; discard
- text buffer when not needed any longer. Fixes #341681.
+2010-03-11 16:09:26 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/textoverlay.c:
- (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
- (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
- (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
- (test_video_waits_for_text_send_text_newsegment_thread),
- (test_video_waits_for_text_shutdown_element),
- (test_render_continuity_push_video_buffers_thread),
- (textoverlay_suite):
- Add some unit tests for textoverlay.
+ * tests/icles/audio-trickplay.c:
+ tests: Fix unitialized variable.
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-11 15:38:18 +0100 Benjamin Otte <otte@redhat.com>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Avoid integer underflow when the found probability for mp3 is
- smaller than the 'penalty' we subtract if there's not a clean
- mp3 header sync at offset 0.
-
-2006-11-21 Stefan Kost <ensonic@users.sf.net>
-
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Add some new symbols to the docs
-
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- * tests/check/elements/ffmpegcolorspace.c:
- (ffmpegcolorspace_suite):
- Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
- (for now not for valgrinding though, since it takes too long).
-
-2006-11-20 Wim Taymans <wim@fluendo.com>
-
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix RGBA32 caps. Fixes #357038.
-
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/interfaces/mixertrack.h:
- Add FIXME so we can add some padding here in 0.11
-
-2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- Fix GstBaseRTPAudioPayload structure so the whole GObject
- inheritance business actually works (parent class instance structure
- must always come first in the derived class instance structure).
-
-2006-11-16 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * ext/ogg/gstoggdemux.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioquantize.h:
+ * gst/audioconvert/gstchannelmix.h:
+ * gst/playback/gstplaysink.c:
+ Add -Wredundant-decls to warning flags
+ ... and fix all the warnings that flag throws.
- * gst/videotestsrc/Makefile.am:
- * tests/check/Makefile.am:
- Make sure our checks and the videotestsrc plugin link against the
- local uninstalled gst libs and not any installed gst libs that
- might happen to exist as well.
+2010-03-11 13:32:14 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/elements/adder.c: (message_received),
- (test_event_message_received), (test_play_twice_message_received):
- * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
- Fix compiler warnings when compiling against core with disabled
- debugging system.
+ * configure.ac:
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstogg.c:
+ * ext/ogg/gstogg.h:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/ogg/vorbis_parse.c:
+ * ext/ogg/vorbis_parse.h:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/gsttheoraenc.h:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/riff/riff.c:
+ * gst-libs/gst/rtsp/gstrtspbase64.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/tag/lang.c:
+ * gst/ffmpegcolorspace/Makefile.am:
+ * gst/ffmpegcolorspace/gstffmpeg.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.h:
+ * gst/gdp/gstgdppay.h:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplayback.c:
+ * gst/playback/gstplayback.h:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/videorate/gstvideorate.h:
+ * tests/check/elements/appsink.c:
+ * tests/check/elements/audiorate.c:
+ * tests/check/elements/audioresample.c:
+ * tests/check/libs/cddabasesrc.c:
+ * tests/check/libs/mixer.c:
+ * tests/check/libs/navigation.c:
+ * tests/examples/gio/giosrc-mounting.c:
+ Add -Wmissing-declarations -Wmissing-prototypes to warning flags
+ Includes all the fixes necessary to make stuff compile again.
-2006-11-16 Michael Smith <msmith@fluendo.com>
+2010-03-11 12:49:02 +0100 Benjamin Otte <otte@redhat.com>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
- (gst_audio_rate_sink_event), (gst_audio_rate_chain):
- Fix audiorate, so that it accurately sets offsets and timestamps.
- Doesn't change the fundamental algorithmic decisions; so should be
- safe.
+ * ext/gio/gstgiobasesink.c:
+ gio: Remove unused function
- * tests/check/Makefile.am:
- Enable audiorate test now that it passes.
+2010-03-11 11:14:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-11-09 Stefan Kost <ensonic@users.sf.net>
+ * ext/vorbis/gstvorbisparse.c:
+ vorbisparse: make sure header buffer metadata is writable before modifying it
+ Fixes unit test failures with core git.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- clear xv when going to NULL, remove // commented non-existant proto
+2010-03-11 12:18:00 +0100 Benjamin Otte <otte@redhat.com>
- * tests/examples/seek/seek.c: (main):
- add missing tooltip description for scrub and play_scrub
+ * tests/check/elements/multifdsink.c:
+ check: Ref buffers after setting caps on them
+ Reffing makes metadata unwritable, so we need to set the caps before.
-2006-11-14 David Schleef <ds@schleef.org>
+2010-03-11 12:04:32 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
- Bump liboil requirement to 0.3.8.
- * gst-libs/gst/riff/riff-media.c:
- Add Dirac fourcc.
- * gst/videoscale/vs_image.h:
- * gst/videoscale/vs_scanline.h:
- Use liboil's stdint.h.
- * gst/videotestsrc/videotestsrc.c:
- Remove liboil related ifdef's, since they aren't needed now, and
- won't work with future versions.
+ Add WARNING_CXXFLAGS where ERROR_CXXFLAGS are
+ This matches the previous commit doing the same for CFLAGS in response
+ to the common/ module changes.
-2006-11-14 David Schleef <ds@schleef.org>
+2010-03-11 12:04:37 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/videoscale/Makefile.am:
- * gst/videoscale/gstvideoscale.c:
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/vs_4tap.c:
- * gst/videoscale/vs_4tap.h:
- * gst/videoscale/vs_image.c:
- * gst/videoscale/vs_image.h:
- * gst/videoscale/vs_scanline.c:
- * gst/videoscale/vs_scanline.h:
- Add a 4-tap image scaler. Theoretically looks much prettier.
- The tap calculation could use some improvement.
-
-2006-11-14 Wim Taymans <wim@fluendo.com>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
+2010-03-11 10:38:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
- (gst_riff_parse_strf_iavs):
- * gst/subparse/gstsubparse.c: (convert_encoding):
- * gst/tcp/gstmultifdsink.c:
- (gst_multi_fd_sink_handle_client_write):
- * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
- (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
- (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Various gsize and gssize printf fixes. Fixes #372507.
-
-2006-11-13 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
- (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
- (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
- (vorbis_dec_chain_forward), (vorbis_dec_chain):
- * ext/vorbis/vorbisdec.h:
- First stab at vorbis reverse playback.
+ * tests/icles/test-xoverlay.c:
+ tests: don't use Gtk+ 2.18 API for no good reason
+ The rest of the code directly uses widget->allocation as well, so no point
+ in using the new API in other places.
-2006-11-13 Wim Taymans <wim@fluendo.com>
+2010-03-11 11:20:48 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Make the clock sync code more accurate wrt resampling and playback
- at different rates.
-
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
- * gst-libs/gst/audio/gstringbuffer.h:
- Use better algorithm to interpolate sample rates.
+ * common:
+ Automatic update of common submodule
+ From df8a7c8 to e272f71
-2006-11-13 Michael Smith <msmith@fluendo.com>
+2010-03-11 10:55:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
- Improve a debug line slightly.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xvoverlay: correct version number in docs
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
- Call gst_riff_init() in plugin_init, to avoid getting errors from
- the debug system (unrelated changes to another plugin made this turn
- up; not sure why).
+2010-02-26 13:56:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-11-10 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/audio-trickplay.c:
+ tests: add a test for trickplay in audio synthesis graphs
+ Right now this mostly demonstatest what not works. That is seeking with
+ start-type = NONE to only update the rate and playing backwards. Also
+ it shows that non-flushing seeks tend to lockup adder. Separate unit tests
+ for the issues follow.
- Patch by: Sergey Scobich <sergery.scobich at gmail com>
+2010-02-08 17:20:35 +0200 Stefan Kost <ensonic@users.sf.net>
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstxmptag.c:
+ * gst-libs/gst/tag/tag.h:
+ * tests/check/libs/tag.c:
* win32/common/libgsttag.def:
- Add missing symbol (#366492).
+ tags: add basic xmp metadata support
+ XMP metadata can be embedded in many media container formats. Implement own
+ parser and formatter that can be used to convert between an xpacket and a
+ GstTagList. Add unit tests.
-2006-11-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
- Don't unref a NULL pad.
+2010-02-19 14:38:36 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-11-09 Wim Taymans <wim@fluendo.com>
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-xoverlay.c:
+ example: add an example for xoverlay::set_render_rectangle()
+ This add a new example which animates a target recangle for the video.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
- (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
- (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
- (gst_ogg_demux_loop):
- Implement first stab at reverse playback.
+2010-02-19 14:46:43 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-11-07 Stefan Kost <ensonic@users.sf.net>
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ xvimagesink: implement set_render_rectangle
+ Previously we hardcoded the target rectangle passes to Xv(Shm)PutImage. Extend
+ the implementation to use a full rectangle and don't assume 0,0 for top,left.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- add h263/h264 variants to the caps, Fixes #363118
+2010-02-17 15:00:13 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-11-06 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ xoverlay: add new vmethod ::set_render_rectangle()
+ Add set_render_rectangle() vmethod to the interface to better support windowless
+ toolkits (e.g. qt graphicsview or video on canvas in general). Right now we
+ always fill the widget to 100%. With the patch we can use a rectangular target
+ region. Fixes #610249.
+ API: GstXOverlay::set_render_rectangle()
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
- * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
- Use g_strerror instead of strerror so we get UTF-8.
+2010-02-16 12:06:08 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-11-03 David Schleef <ds@schleef.org>
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: take new size from event thread and do not poll for every frame
+ We can update the geometry in ConfigureNotify (unless we disable event-
+ handling). If event handling is disabled, one should use _expose() to trigger a
+ redraw and update the geometry.
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- Add/remove KW-DIRAC header here, since it is ogg-specific.
+2010-03-10 21:51:59 +0100 Benjamin Otte <otte@redhat.com>
-2006-11-03 Michael Smith <msmith@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 9720a7d to df8a7c8
- * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
- Recognise more mpeg4 elementary video streams.
+2010-03-10 21:01:20 +0100 Benjamin Otte <otte@redhat.com>
-2006-11-02 Edward Hervey <edward@fluendo.com>
+ * configure.ac:
+ Update for recent changes to common submodule
+ This just replaces every "$ERROR_CFLAGS" usage with a usage of
+ "$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
+ previously.
+ Actually using that separation will happen later.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Lower the probability of mp3 typefinding functions if we don't find a
- valid mp3 header at the start of the file.
- Closes #369482
+2010-03-10 20:43:46 +0100 Benjamin Otte <otte@redhat.com>
-2006-11-02 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 0b6e072 to 9720a7d
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (theora_dec_sink_event), (theora_dec_chain_forward),
- (theora_dec_flush_decode), (theora_dec_chain_reverse),
- (theora_dec_chain):
- Document and partially implement an algorithm for doing reverse playback
- of theora video.
+2010-03-10 16:09:45 +0100 Benjamin Otte <otte@redhat.com>
-2006-11-02 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 7cc5eb4 to 0b6e072
- Patch by: Sergey Scobich <sergey.scobich at gmail com>
+2010-03-10 14:36:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * win32/common/config.h:
- * win32/common/interfaces-enumtypes.c:
- * win32/common/libgsttag.def:
- * win32/vs8/gst-plugins-base.sln:
- * win32/vs8/libgstaudioresample.vcproj:
- * win32/vs8/libgstinterfaces.vcproj:
- * win32/vs8/libgstogg.vcproj:
- * win32/vs8/libgstriff.vcproj:
- * win32/vs8/libgsttag.vcproj:
- * win32/vs8/libgsttheora.vcproj:
- * win32/vs8/libgstvideoscale.vcproj:
- * win32/vs8/libgstvorbis.vcproj:
- Misc. VS8 build fixes: fix syntax in config.h, add missing entries
- to libgsttag.def; add missing dependencies for some vs8 projects;
- re-arrange placement of .def files in vs8 projects (#366334).
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: do not cache FLUSH_START/_STOP events
+ ... and similarly so for serialized events.
-2006-11-01 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 14:34:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/ogg/gstogg.c:
- Remove unused variable.
+ * gst/playback/gstplaysink.c:
+ playsink: provide correct error message if configured audio/video sink fails
- * ext/ogg/gstoggdemux.c:
- Fix Wim's surname in plugin description.
+2010-03-10 10:22:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-10-31 Wim Taymans <wim@fluendo.com>
+ * ext/vorbis/gstvorbisdec.h:
+ vorbisdec: remove unused field
- * gst-plugins-base.spec.in:
- spec new .h file. Fixes #368310.
+2010-02-02 11:34:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-10-31 Michael Smith <msmith@fluendo.com>
+ * tests/check/pipelines/vorbisdec.c:
+ tests: enable strict discontinuity checking on vorbisdec pipeline
+ Closes #423086.
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
- (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
- (gst_multi_fd_sink_get_stats),
- (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_queue_buffer),
- (gst_multi_fd_sink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Make using the remove or clear signals threadsafe.
- Make calling get-stats with an invalid fd not segfault.
- Fixes 368273.
+2010-03-10 01:09:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-31 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 7aa65b5 to 7cc5eb4
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_init):
- Fix and activate base audio payloader.
+2010-03-10 01:07:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/video/video.c:
+ docs: fix Returns: for gst_video_parse_caps_color_matrix()
- * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
- (plugin_init):
- Add typefinder for QuickTime Image Files (see #366156).
+2010-03-10 00:46:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update for changed string
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Another typo fix (#366212).
+2010-03-10 00:42:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-27 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/videorate.c:
+ tests: fix typo in videorate unit test pipeline description
+ Two consecutive ! ! leave a 'Link without source' error in the debug log.
- * gst/volume/gstvolume.c: (volume_transform_ip):
- Use stream time to synchronize volume property instead of rather random
- timestamps. This is needed when gnonlin does its time shifting.
+2010-03-10 00:41:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-27 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/videorate.c:
+ tests: don't use deprecated functions in videorate unit test
- Patch by: Mark Nauwelaerts <manauw at skynet dot be>
+2010-03-10 00:29:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
- Remove the pad from the element in release_pad. Fixes #364812.
+ * win32/common/libgstvideo.def:
+ win32: add new API to libgstvideo.def
-2006-10-27 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-09 15:39:55 -0800 David Schleef <ds@schleef.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
- Explicitly create our custom buffer classes at a thread-safe
- location as well, since g_type_class_ref() doesn't seem to be
- entirely thread-safe either (#365501; also see #349410).
+ * ext/ogg/gstoggmux.c:
+ oggmux: Don't flush after every frame for theora
-2006-10-26 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-09 21:26:58 +0000 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
- (gst_riff_parse_info):
- If strings in INFO chunk are not UTF-8, do something similar to
- what we do for ID3v1 tags: check a number of environment variables
- (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
- character sets to try, otherwise try the current locale and/or fall
- back on ISO-8859-1. Fixes #360552.
+ * common:
+ Automatic update of common submodule
+ From 44ecce7 to 7aa65b5
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-09 13:05:23 -0800 David Schleef <ds@schleef.org>
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_pattern_get_type),
- (gst_video_test_src_set_pattern):
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
- (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
- (gst_video_test_src_checkers8):
- * gst/videotestsrc/videotestsrc.h:
- Add a bunch of exciting new checkers patterns.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add color-matrix handling to caps
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-30 22:55:01 -0800 David Schleef <ds@schleef.org>
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (handle_buffer),
- (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
- * gst/subparse/gstsubparse.h:
- * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
- (parse_tmplayer):
- * gst/subparse/tmplayerparse.h:
- Add support for TMPlayer-type subtitles (#362845).
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Add color-matrix to caps
- * tests/check/elements/subparse.c: (test_tmplayer_do_test),
- (GST_START_TEST), (subparse_suite):
- Add some basic unit tests for the above.
+2010-02-26 16:25:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ * pkgconfig/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tools/Makefile.am:
+ build: Make some more rules silent if requested
- * tests/check/elements/audiorate.c: (test_injector_base_init),
- (test_injector_class_init), (test_injector_chain),
- (test_injector_init), (probe_cb), (do_perfect_stream_test),
- (GST_START_TEST), (audiorate_suite):
- More tests for audiorate: inject buffers to check behaviour when
- buffers overlap.
+2010-02-26 15:40:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-21 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure: Use automake 1.11 silent rules instead of shave if available
+ This makes sure that we use something that is still maintained and
+ also brings back libtool 1.5 support.
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
- (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
- Add some basic unit tests for audiorate. Disabled at the moment
- since it doesn't pass yet (see bug #363119).
+2010-02-23 19:12:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Don't fail if there are subtitles and audio but no video
+ Change playbin2 to not error out if there are subtitles and audio
+ but no video. If visualizations are enabled the subtitles are rendered on top
+ of the visualization stream, otherwise the subtitles are not linked at all and
+ only the audio is played (and a warning message is posted).
+ If there are only subtitles but neither audio nor video an error message is
+ still posted.
+ Fixes bug #610866.
- * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
- (parse_subrip), (handle_buffer):
- Add missing closing tags for markup and fix broken markup,
- otherwise pango won't render anything (fixes #357531). Also,
- make sure the text we send out is always NUL-terminated
- (better safe than sorry etc.).
+2010-02-17 19:18:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/subparse.c: (test_srt_do_test),
- (test_srt):
- Some more tests for .srt incl. tests for the above stuff.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ volume: If a controller is used, use sample accurate property values
+ Fixes bug #609801.
-2006-10-20 Julien MOUTTE <julien@moutte.net>
+2010-03-09 19:17:04 +0100 Benjamin Otte <otte@redhat.com>
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
- Patch by: Stefan Kost <ensonic@users.sf.net>
- Try to redraw borders only when needed. Apparently this consumes
- resources on small devices... :-O (#363607)
+ * gst-libs/gst/video/video.c:
+ gstvideo: Fix typos in comments
-2006-10-20 Michael Smith <msmith@fluendo.com>
+2010-03-09 17:32:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c:
- (gst_multi_fd_sink_client_queue_buffer):
- If caps change, then update the client's idea of the caps so that we
- don't end up re-sending streamheaders for every single buffer after
- the caps change.
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development
-2006-10-20 Michael Smith <msmith@fluendo.com>
+=== release 0.10.28 ===
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
- (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
- Set caps on pushed buffers; fix up refcounting of caps objects.
+2010-03-08 23:20:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.28
- * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
- (plugin_init):
- Typefind mmsh header data packet to application/x-mmsh (#362625).
+2010-03-08 23:19:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/subparse.c: (buffer_from_static_string),
- (setup_subparse), (teardown_subparse), (test_srt_do_test),
- (GST_START_TEST), (subparse_suite):
- Add very simple unit test for subparse.
+2010-03-08 21:57:03 +0100 Benjamin Otte <otte@redhat.com>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoraenc.c:
+ theora: Fix SIGFPE when using 0/1 framerate
+ libtheora crashes with a 0 framerate, so let's forbid it.
+ https://bugzilla.redhat.com/show_bug.cgi?id=571289
- * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
- (parse_subrip):
- Strip trailing newlines from subtitle text output.
+2010-03-08 14:50:25 +0000 David Schleef <ds@schleef.org>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/dirac_parse.c:
+ oggdemux: fix dirac header parsing
+ Fixes #611900.
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_change_state):
- Fix memleak; clear subparse->textbuf n state change function.
+2010-03-08 14:46:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/overlay/Makefile.am:
+ examples: make sure to dist qtgv-xoverlay.h header file
+ This time for real.
+ Fixes #610832.
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't require subrip (.srt) files to start with a chunk number of 1.
+2010-03-08 12:11:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-18 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basedepay: clarify some documentation
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Extract rate from the NEWSEGMENT event.
- Use commit_full to also take rate adjustment into account when writing
- samples to the ringbuffer.
-
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- Added _commit_full() to also take rate into account.
- Use simple interpolation algorithm to resample audio.
- API: gst_ring_buffer_commit_full()
+2010-03-08 11:25:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
- * tests/examples/seek/seek.c: (segment_done):
- Don't try to seek with 0.0 rate, just pause instead.
- Remove bogus debug line.
+ * ext/alsa/gstalsasrc.c:
+ alsasrc: return right number of bytes that we wrote
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-08 11:20:51 +0100 Dake Gu <gudake@gmail.com>
- * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
- (setup_source):
- Catch async errors when starting up the subtitle bin, so we can
- stop waiting and continue with the main film instead of hanging
- forever. Fixes #339366.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: fix handling of x-server-ip-address
+ Fix handling of x-server-ip-address.
- * tests/check/elements/playbin.c: (playbin_suite):
- Enable unit test for the above.
+2010-03-02 11:25:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/design/draft-keyframe-force.txt:
+ docs: update keyframe force event
+ Add field to send all headers.
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/playbin.c: (GST_START_TEST),
- (gst_red_video_src_uri_get_type),
- (gst_red_video_src_uri_get_protocols),
- (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
- (gst_red_video_src_uri_handler_init),
- (gst_red_video_src_init_type), (gst_red_video_src_base_init),
- (gst_red_video_src_create), (gst_red_video_src_class_init),
- (gst_red_video_src_init), (plugin_init), (playbin_suite):
- Some small and basic unit tests for playbin; not very useful yet,
- but at least a start.
+=== release 0.10.27 ===
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-06 00:09:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybin.c: (setup_sinks):
- The old pad activation spiel.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.27
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-06 00:08:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Don't hang forever if the subbin already fails to start up in
- the state change to PAUSED (#339366).
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
-2006-10-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-05 15:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
- (gst_tuner_set_channel), (gst_tuner_get_channel),
- (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
- (gst_tuner_set_frequency), (gst_tuner_get_frequency),
- (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
- (gst_tuner_find_channel_by_name):
- Fix some function guards, add some more function guards.
+ * configure.ac:
+ configure: first check for QtGui >= 4.6, only then for >= 4.0
+ If we first check for >= 4.0 the second check for >= 4.6 will just
+ short-cut since we are using the same prefix for the variables for
+ both checks, and they've already been set previously. So the examples
+ requiring >= 4.6 were built even in the >= 4.0 case.
-2006-10-17 Jan Schmidt <thaytan@mad.scientist.com>
+2010-03-03 20:18:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
- (remove_element_chain):
- Don't return a pad from get_our_ghost_pad unless it is actually the
- one we want.
- Change a cast in remove_element_chain slightly.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.4 pre-release
-2006-10-13 Julien MOUTTE <julien@moutte.net>
+2010-03-03 20:17:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek), (start_seek),
- (rate_spinbutton_changed_cb), (segment_done), (msg_state_changed):
- Segment seeking needs to use the rate and set stop to -1.
+ * po/ja.po:
+ po: update translations
-2006-10-13 Wim Taymans <wim@fluendo.com>
+2010-03-03 20:15:44 +0000 Josep Torra Valles <n770galaxy@gmail.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_setcaps):
- Don't crash when ringbuffer is not yet created.
- Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
- Fixes #361634.
+ * gst/playback/gstplaysink.c:
+ playsink: avoid g_object_set() on NULL pointers
+ There may not be an overlay element if a text-sink is set.
+ Fixes #611702.
- * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_request_new_pad):
- Activate pads befre adding them to running elements.
+2010-03-01 12:17:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-13 Julien MOUTTE <julien@moutte.net>
+ * ext/ogg/gstoggstream.c:
+ oggstream: mark skeleton streams correctly
+ Mark skeleton streams because we need to ignore them for calculating the
+ duration of the stream.
+ Fixes #611227
- * tests/examples/seek/seek.c: (do_seek), (start_seek),
- (rate_spinbutton_changed_cb), (msg_state_changed): Stop the scale
- updater when we start grabing the slider. Don't wait for the
- pipeline to be PAUSED.
+2010-02-24 01:10:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-13 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * po/nl.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.3 pre-release
- * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
- (gst_mixer_set_volume), (gst_mixer_get_volume),
- (gst_mixer_set_mute), (gst_mixer_set_option),
- (gst_mixer_get_option), (gst_mixer_mute_toggled),
- (gst_mixer_record_toggled), (gst_mixer_volume_changed),
- (gst_mixer_option_changed):
- Guard mixer interface functions against bogus arguments.
+2010-02-23 16:57:53 +0100 Götz Waschk <waschk@mandriva.org>
-2006-10-12 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/overlay/Makefile.am:
+ examples: Dist header file for the Qt graphics view example
+ Fixes bug #610832.
- * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
- (play_cb), (pause_cb), (stop_cb), (rate_spinbutton_changed_cb),
- (msg_state_changed), (main): Use state-changed messages to trigger
- start/stop of scale update timer. Indeed the scale slider was
- jumping here and there because the update timer was activated
- before seek completed. This fixes instant applying of rate changes
- by pressing the spinbutton like a crazy man !
+2010-02-23 11:41:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-12 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use the chain begin_time instead of our counter
+ We update the passed begintime argument to narrow our search region in the
+ binary search. This means that it does not always contain the chain begin time
+ after a couple of bisects. Use the real chain->begin_time to bring the
+ granuletime to the time in the chain instead.
+ Fixes #610005
- Patch by: Sebastien Cote <sebas642 at yahoo.ca>
+2010-02-19 18:24:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_finalize):
- Fix two small memory leaks (#361456).
+ * tests/check/elements/videorate.c:
+ videorate: tests: New unit tests for upstream caps nego
+ Adds unit tests that check videorate's upstream caps
+ negotiation works properly (put passthrough caps
+ first)
+ Fixes #608025
-2006-10-10 Julien MOUTTE <julien@moutte.net>
+2010-01-27 15:07:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek),
- (rate_spinbutton_changed_cb): When changing spinbutton we try
- to change the rate on the fly.
+ * gst/videorate/gstvideorate.c:
+ videorate: Improve upstream negotiation
+ Put peer pad caps preferred framerates first, indicating
+ they are videorate's first choices, removing an unnecessary
+ conversion.
+ Fixes #608025
-2006-10-10 Wim Taymans <wim@fluendo.com>
+2010-02-21 19:52:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Add WMS caps.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
+ For this add subtitle encoding properties to playsink and subtitleoverlay
+ and update the values in the containing elements.
+ Also update the font description in textoverlay or the used renderer
+ element if it is changed during playback.
+ Fixes bug #610310.
-2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2010-02-22 13:01:19 +0200 Stefan Kost <ensonic@users.sf.net>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/examples/overlay/qt-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.cpp:
+ examples: also add sink detection and set title to qt examples
+ Also set a title in the qt examples like it is now done in the gtk example.
+ Fix the newly added find_video_sink in the gtk example and add similar function
+ to the qt examples.
- * ext/gnomevfs/gstgnomevfssink.c:
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix URI interface implementation return type.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
- Fix what looks like a copy/paste issue when assigning values.
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_get_type):
- Cast to prevent Forte warnings.
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
- Fix URI interface implementation return type.
- gst_pad_query_position requires a signed integer pointer as
- 3rd parameter, GstClockTime is unsigned.
- * gst/audioconvert/audioconvert.c:
- Fix integer overflow when treated as signed.
- * gst/audioresample/resample.c: (resample_add_input_data):
- Cast to prevent warnings on Forte.
- * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
- Fix integer overflow when treated as signed.
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
- * gst/playback/gstdecodebin.c: (queue_filled_cb),
- (cleanup_decodebin):
- Who initialises a guint to -1!
- Cast function pointers to prevent warnings on Forte.
- * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
- (queue_threshold_reached):
- Cast function pointers correctly to prevent warnings on Forte.
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
- Cast function pointers correctly to prevent warnings on Forte.
- * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
- Obvious change to unsigned, 0xEF > max signed char.
- * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
- GstClockTime is unsigned, initialise correctly.
- * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
- Cast so pointer arithemetic doesn't cause warnings on Forte.
- * gst/videorate/gstvideorate.c:
- Use correct return value.
- * tests/examples/seek/scrubby.c:
- GstClockTime is unsigned, initialise correctly.
+2010-02-19 14:40:43 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-10-10 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/overlay/.gitignore:
+ gitignore: ignore files in new example directroy
- Patch by: Ferenc Gerlits <fgerlits at gmail com>
+2010-02-17 14:59:33 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/typefind/gsttypefindfunctions.c:
- Recognise XML files and XML-like files shorter than 256 bytes as
- well (fixes #359237).
+ * gst-libs/gst/video/Makefile.am:
+ make: fix copy and paste error in git rules (audio<->video)
-2006-10-09 Edgard Lima <edgard.lima@indt.org.br>
+2010-02-19 17:44:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Renato Filho <renato.filho@indt.org.br>
-
- * gst/typefind/gsttypefindfunctions.c:
- Added typefind functions to video/x-nuv media.
-
-2006-10-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Ghost the video sinkpad if a text sinkpad is available
+ Only don't ghost it if no visualizations are need and if
+ no text is needed and no textchain was created yet.
+ Fixes bug #610379.
- * gst-libs/gst/interfaces/xoverlay.c:
- (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
- Some more guards against invalid input.
+2010-02-19 00:22:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-07 Julien MOUTTE <julien@moutte.net>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.2 pre-release
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
- Useless goto.
- * tests/examples/seek/seek.c: (do_seek),
- (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
- seek example to experiment with rates != 1.0 (reverse playback !)
+2010-02-19 00:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-06 Stefan Kost <ensonic@users.sf.net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translation files
- * gst-libs/gst/interfaces/xoverlay.c:
- Unref message in doc-example (spotted by Robert McQueen)
+2010-02-19 00:17:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * tests/examples/overlay/.gitignore:
+ Ignore new overlay examples
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- printf fix.
+2010-02-18 23:47:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * tests/examples/overlay/gtk-xoverlay.c:
+ examples: don't hard-code xvimagesink for Gtk+ GstXOverlay example
+ Try to find a working videosink, don't hardcode xvimagesink. Also
+ add some borders to window and give it a title so that it's clear
+ that this is really a Gtk+ window and not a window created by the
+ videosink.
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (close_pad_link):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
- Activate dynamic pads before adding them to the element.
+2010-02-18 11:42:55 -0800 David Schleef <ds@schleef.org>
-2006-10-06 Michael Smith <msmith@fluendo.com>
+ * gst/tcp/gsttcp.c:
+ tcp(client/server)src: Fix handling of closed sockets
+ The peer closing the socket should cause an EOS, instead of
+ silently doing nothing. This changes the behavior to be
+ more like fdsrc. Fixes: #610386
- * gst-libs/gst/floatcast/floatcast.h:
- Fix obviously-bogus macros; use the correct types.
+2010-02-18 12:42:53 +0000 Patrick Radizi <patrick.radizi@axis.com>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: make sure not to dereference NULL username or password
+ Fixes #610268.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_change_state):
- Also call parent state change function to activate pads.
+2010-02-17 21:22:54 -0800 David Schleef <ds@schleef.org>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- Add some more debug info in mpeg typefinding.
+ * ext/theora/gsttheoradec.c:
+ theoradec: Fix chroma copying for 4:2:2
+ Fix mixup of height/width, causing only half the chroma lines to
+ be copied when outputting buffers. Fixes: #610329.
-2006-10-06 Michael Smith <msmith@fluendo.com>
+2010-02-16 15:43:26 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/theora/theoradec.c: (theora_dec_chain):
- Zero byte theora packets are valid and well-defined; don't warn on
- them.
+ * configure.ac:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/examples/overlay/qt-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.h:
+ examples: add video overlay examples for gtk, qt and qt graphics view
+ Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
+ adds all boilerplate to configure for using c++. The qt based examples are
+ optional like their gtk counterparts.
+
+2010-02-16 17:20:01 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/libs/compiling.sgml:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: cleanup library docs
+ Correct name of included files. Remove files that are not used anymore. Add many
+ new api entries to their sections.
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_get_stats), (find_limits),
- (gst_multi_fd_sink_queue_buffer):
- API: add dropped_buffers to the get-stats GValueArray
+2010-02-15 11:11:04 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-10-05 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/test-colorkey.c:
+ test-colorkey: remove the XInitThreads()
+ We don't do this is any other example, this should be done for us in gdk it if
+ would be needed.
- * ext/alsa/gstalsadeviceprobe.c:
- (gst_alsa_device_property_probe_get_values):
- * ext/alsa/gstalsasink.c: (set_hwparams):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
- (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
- (gst_ogg_mux_process_best_pad):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
- (gst_ogg_parse_chain):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
- (gst_vorbis_enc_buffer_check_discontinuous):
- * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_handle_track_seek):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_push_full):
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- * gst/audioresample/resample.c: (resample_input_pushthrough):
- * gst/playback/gstplaybasebin.c: (queue_out_of_data):
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (wavpack_type_find):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
- * tests/check/elements/volume.c: (GST_START_TEST):
- Printf format fixes.
+2010-02-16 10:09:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use same message string for missing elements as in playbin
+ Use the same translated message string for missing core elements as
+ playbin uses, which is a bit nicer and also indicates that there is
+ something wrong with the user's GStreamer installation (which arguably
+ is the case if elements like typefind or queue2 are missing).
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
- Fix a simple mistake (see the docs)
- Fixes #359580
+2010-02-08 13:54:57 +0200 Kaj-Michael Lang <milang@tal.org>
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Handle stm module format
+ Fixes #609314.
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-cdparanoia.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gdp.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playbin.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml:
- Add vorbistag element to docs; update version numbers to 0.10.10.1.
+2010-02-15 12:10:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/gstivorbisdec.c:
+ ivorbisdec: set rank to SECONDARY
- Patch by: James "Doc" Livingston <doclivingston at gmail com>
+2010-02-15 12:09:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * configure.ac:
+ * ext/Makefile.am:
* ext/vorbis/Makefile.am:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
- (vorbis_parse_parse_packet), (vorbis_parse_chain):
- * ext/vorbis/vorbisparse.h:
- * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
- (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
- (gst_vorbis_tag_parse_packet):
- * ext/vorbis/vorbistag.h:
- Add new vorbistag element which derives from vorbisparse
- and is essentially the same as well, only that it implements
- the GstTagSetter interface and can modify the stream's
- vorbiscomment on the fly (#335635).
-
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/vorbistag.c: (setup_vorbistag),
- (cleanup_vorbistag), (buffer_probe), (start_pipeline),
- (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
- (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
- Add unit test for new vorbistag element.
-
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/gstivorbisdec.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/gstvorbisdeclib.c:
+ * ext/vorbis/gstvorbisdeclib.h:
+ vorbisdec: also support ivorbis tremor decoder
+ ... which only needs a bit of refactoring and extracting to support
+ the minor difference in (i)vorbis interface.
+ Fixes #609063.
- * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
- (vorbis_parse_push_headers), (vorbis_parse_chain):
- Set BOS flag in packet structure to fix 'jump depends
- on unitialized value' errors in valgrind; various minor
- clean-ups.
-
-2006-09-30 Jan Schmidt <thaytan@mad.scientist.com>
+2010-02-03 14:37:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix typo in a debug statement.
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisdec.h:
+ vorbisdec: reduce some hard-coding
+ ... such as assuming float all over, and base src caps on template caps.
- * gst/playback/gstplaybasebin.c: (probe_triggered),
- (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
- (gen_source_element), (source_new_pad), (analyse_source),
- (setup_source):
- When handling no_more_pads in new_decoded_pad, make sure to treat
- subtitle pads correctly. Fixes playback with subtitle files.
+2010-02-15 10:23:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Move a recurring message to LOG level.
+ * tests/check/elements/playbin.c:
+ playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
- which ends up as -1 when cast to an int. Make the logic handle the
- max value as an unsigned mask and only change the colorkey when it's
- a value we recognise.
+2010-02-15 09:04:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * ext/ogg/gstoggparse.c:
+ oggparse: Fix another format string compiler warning
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- Removed empty * between paragraphs
+2010-02-15 08:56:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix format string compiler warnings
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/README:
- Moved some documentation into .c file
+2010-02-15 08:48:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Post a missing element message and an error message if no uridecodebin can be found
- * gst/playback/gstdecodebin.c: (no_more_pads):
- Fix compilation.
+2010-02-15 08:46:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playsink: Post missing element messages if a core plugin is missing
+ And post a warning in cases where we can still continue to work
+ or an error when the missing element is fatal.
- * gst/playback/gstdecodebin.c: (new_caps):
- Remove g_print
+2010-02-15 08:28:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Add some docs.
+ * tests/check/elements/playbin2.c:
+ playbin2: Enable all unit tests
+ They're all working and valgrind clean now.
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 08:26:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/Makefile.am:
- Re-enable cddabasesrc test to see if it works again
- now.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: First post a missing-plugin message, then emit the unkown-type signal
+ This makes sure that there *always* is a missing plugin message in the bus
+ before any errors or warning messages.
-2006-09-29 Wim Taymans <wim@fluendo.com>
+2010-02-15 08:20:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (setup_subtitle),
- (gen_source_element):
- Handle invalid URIs a bit more gracefully.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
+ and not CORE MISSING_PLUGIN.
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 08:18:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/oggmux.c:
- Remove obsolete comment.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Free the subtitle URI
-2006-09-29 Michael Smith <msmith@fluendo.com>
+2010-02-15 08:06:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
- (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
- (gst_ogg_mux_collected):
- Commit patch from James "Doc" Livingston, adds proper EOS handling
- in oggmux. GStreamer can, for the first time ever, create a valid
- Ogg file! Yay!
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post missing plugin messages if a required element can't be created
+ Especially if no suitable URI source can be found.
- * tests/check/pipelines/oggmux.c: (check_chain_final_state),
- (oggmux_suite):
- Reenable tests now that they pass.
+2010-02-15 06:50:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
- Stop reading commands when EOF (we read 0) as well.
-
-2006-09-28 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
- (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
- (find_dynamic), (unlinked), (close_link):
- Implement delayed caps linking needed for element with a lot of
- different caps on the src pads that get fixed at runtime.
- Improve management of dynamic elements.
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (group_destroy), (group_commit), (check_queue), (queue_overrun),
- (gen_preroll_element), (remove_groups), (unknown_type),
- (add_element_stream), (no_more_pads_full), (no_more_pads),
- (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
- (new_decoded_pad), (setup_subtitle), (array_has_value),
- (gen_source_element), (source_new_pad), (has_all_raw_caps),
- (analyse_source), (remove_decoders), (make_decoder),
- (remove_source), (setup_source), (finish_source), (prepare_output),
- (gst_play_base_bin_change_state):
- * gst/playback/gstplaybasebin.h:
- Use more _CAST instead of full type checking casts.
- Small cleanups, plug some leaks.
- Handle dynamic sources.
- Add some helper functions to create lists of strings used for
- blacklisting and other stuff.
- Refactor some code dealing with analysing the source.
- Re-enable sources without pads (like cd:// or other selfcontained
- elements).
+ * tests/check/elements/.gitignore:
+ tests: Add decodebin2 test to .gitignore
-2006-09-28 Wim Taymans <wim@fluendo.com>
+2010-02-15 01:18:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- When we have a timestamp, we can still perform clipping.
- When we have no clock, we must play the sample ASAP.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Set ghostpad targets to NULL when freeing a decode chain
+ Otherwise the ghostpad will still be linked to the peer and there
+ will still be a reference kept, leading to nothing being unlinked
+ and destroyed until decodebin2 is finalized.
+ This fixes reuse of decodebin2 if a raw stream is connected to
+ its sinkpad.
-2006-09-28 Wim Taymans <wim@fluendo.com>
+2010-02-15 01:17:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Set caps on outgoing buffers.
+ * tests/check/Makefile.am:
+ * tests/check/elements/decodebin2.c:
+ decodebin2: Add simple unit test, mainly a copy of the decodebin unit test
+ The only difference between the two unit tests right now is,
+ that the decodebin2 test resets the element to READY before trying
+ to reuse it instead of NULL. decodebin2 guarantees to be reusable
+ without going back to NULL.
- * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
- (gst_video_rate_event), (gst_video_rate_chain):
- * gst/videorate/gstvideorate.h:
- Fix videorate some more. Fixes #357977
+2010-02-15 00:11:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
+ See #609252.
- * tests/check/elements/adder.c: (adder_suite):
- Don't set timeout to 6 seconds when we're running
- in valgrind ... (and how is 6 seconds longer than
- the default anyway?)
+2010-02-14 23:16:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-28 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 96dc793 to 44ecce7
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
- (gst_audio_rate_sink_event), (gst_audio_rate_convert),
- (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
- Keep sink and src segment to keep track of time and support more
- input formats.
- Fix bogus next_offset and run_time calculation, don't understand how
- this could have worked before. Fixes #357976.
- Remove some unneeded vars.
+2010-02-14 23:10:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/Makefile.am:
+ playbin2: Enable playbin2 unit test
+ It now contains a single working unit test and can be enabled.
+ The other more useful unit tests still need fixing.
- * gst/playback/gstplaybin.c: (remove_sinks):
- Only remove visualisation from visbin if there is a visbin (or:
- don't throw warnings when closing totem without playing a file).
+2010-02-14 22:16:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-27 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/playbin.c:
+ playbin: Fix indention in the unit test
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Add some more info in a WARNING.
+2010-02-13 01:08:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Handle PAUSE in create function, use new -core addition to
- wait for playing. Fixes pausing and resuming capture from an
- audiosrc.
+ * gst/volume/gstvolume.c:
+ volume: Replace this variables by self
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Constify some more.
- Caller supports interrupted reads now.
+2010-02-12 19:43:13 +0100 Josep Torra Valles <n770galaxy@gmail.com>
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
+ This makes sure that we don't destroy the last reference before the
+ element gets back to NULL state. Fixes assertion failures if a playbin2
+ instance is reused but different sinks are automatically chosen because
+ of different caps.
- * tests/check/Makefile.am:
- Another attempt to make the gen64 buildbot happy.
+2010-02-12 18:00:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-27 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix Since tag
- Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
+2010-02-12 14:19:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_clear_actors),
- (gst_visual_chain), (gst_visual_change_state):
- Libvisual plugin was not passing audio data to libvisual 0.4.0
- correctly. Fixes #357800
+ * gst-libs/gst/riff/riff-read.c:
+ riff: treat JUNQ chunks like JUNK chunks
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-12 14:29:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
- Add timeout to _get_state() so we see which pipeline it is
- that causes trouble on the gen64 build bot.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update basesrc segment duration and post duration messages from the streaming thread
-2006-09-27 Wim Taymans <wim@fluendo.com>
+2010-02-11 14:10:02 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
- (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
- (gst_base_rtp_depayload_set_gst_timestamp):
- the source pad always uses fixed caps.
+ * gst-libs/gst/tag/tags.c:
+ tags: improve docs about determining the encoding
-2006-09-27 Wim Taymans <wim@fluendo.com>
+2010-02-11 14:09:05 +0200 Stefan Kost <ensonic@users.sf.net>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudioclock.c:
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiosink.c:
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstaudiosrc.c:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.h:
- Added docs for the audio libs.
+ * gst-libs/gst/tag/gstvorbistag.c:
+ comment: fix wrong header comment
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-01 13:50:14 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/check/Makefile.am:
- Temporarily disable test that fails on the bots for unknown reasons.
+ * gst-libs/gst/riff/riff-ids.h:
+ riff: add a variant of the JUNK tag that several adobe products produce
+ JUNQ has same semantics as JUNK.
-2006-09-26 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+2010-02-01 19:01:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- Moved AudioCodecType into priv
- Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
-
-2006-09-25 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
- (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
- (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
- (new_pad):
- Cleanups and small leak fixes.
- Added Depayloaders to valid list of autopluggable elements.
-
-2006-09-25 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
- (gen_video_element), (gen_text_element), (gen_audio_element),
- (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
- (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
- Detect NO_PREROLL state change returns and disable clock distribution to
- the sinks so that sync is disabled.
- Avoid some type checking and do simple casts instead.
- Small cleanups, fix some FIXMEs.
- Be more robust when linking user specified elements, catch an report
- errors. Fixes #357404.
- Fix some leaks in the error paths.
-
-2006-09-25 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: add min-percent property
+ Emit need-data when the amount of data in the internal queue drops below
+ min-percent.
+ Fixes #608309
- * ChangeLog:
- ChangeLog surgery for missing bug-number
+2010-02-01 18:56:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: cleanups
+ Avoid some typechecks.
+ Avoid dereferencing appsrc->priv all the time.
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2010-02-01 18:55:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/test.c:
- Fix compilation with uClibc and -Werror (#357591).
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: cleanups
+ Avoid some typecasting.
+ Avoid dereferencing appsink->priv all the time.
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-01 15:09:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Parse dates that are followed by a time as well (#357532).
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: avoid some typecasts
- * tests/check/libs/tag.c: (test_vorbis_tags):
- Add unit test for this.
+2010-01-29 16:34:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: ignore \n and \r as the first line
+ Be more forgiving for bad servers and ignore \r and \n when we are looking for
+ the response/request line.
+ See #608417
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
- (gst_audio_convert_transform_caps):
- * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
- * gst/videotestsrc/videotestsrc.h:
- A few array const-ifications.
+2010-02-10 16:05:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fail gracefully on bad Content-Length headers
+ Be careful when allocating the amount of bytes specified in the Content-Length
+ because it can be an insanely huge value. Try to allocate the memory but fail
+ gracefully with a nice error when the allocation failed.
- * tests/check/Makefile.am:
- See if this makes the build bots happy.
+2010-02-10 10:12:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/cddabasesrc.c:
- UTF8-ise my name.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-09 17:39:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update segment duration and post a duration message if the duration changes
+ Fixes bug #609423.
- * gst/subparse/samiparse.c: (handle_start_font),
- (fix_invalid_entities):
- More case-insensitivity for certain tags; recognise entities with
- decimal codes as special entities as well (#357330).
+2010-02-11 10:56:17 +0100 Benjamin Otte <otte@redhat.com>
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/Makefile.am:
+ build: link to libm in examples that use it
+ This fixes build failure in Fedora 13.
- * gst-libs/gst/Makefile.am:
- Need to build tag directory before cdda.
+2010-02-11 01:11:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+ * MAINTAINERS:
+ Update MAINTAINERS, add myself
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_base_init):
- * gst-libs/gst/cdda/gstcddabasesrc.h:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
- (gst_tag_register_musicbrainz_tags):
- Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
- depend on libgsttag. This is required so we can extract/read tags like
- DISCID without depending on libgstcddabasesrc (which used to register
- them).
+2010-02-11 23:57:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c:
- Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
- tags (also see #347848).
+ * configure.ac:
+ configure: back to development
+ Slushy freeze remains in effect.
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
- Log vorbis comments we are actually writing. Const-ify array.
+=== release 0.10.26 ===
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2010-02-10 20:17:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gen_preroll_element):
- Improve buffering a bit by avoiding a deadlock because we cannot assume
- the underrun is always called.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.26
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2010-02-10 20:16:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Added MPEG-4 AAC and id and caps. Fixes #357289
- Added WMA9 Lossless id.
+2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
-2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoradec.c:
+ theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
+ Fixes #609252.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix misleading docs addition.
+2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
- * tests/check/elements/videotestsrc.c: (check_rgb_buf):
- Get rid of compiler warning the right way.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: use the default granpos functions for kate streams
+ Set timestamps on kate packets. See bug #600929.
-2006-09-22 Wim Taymans <wim@fluendo.com>
+2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_finalize),
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_push_full),
- (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_process),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Small cleanups.
- Fix some leaks.
- Refactored the process method and added methods to push from the process
- vmethod.
- Use _scale functions.
- API: gst_base_rtp_depayload_push_ts
- API: gst_base_rtp_depayload_push
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.25.3 pre-release
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- timestamps are uint.
+2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-22 Stefan Kost <ensonic@users.sf.net>
+ * po/bg.po:
+ po: update translations
- * gst-libs/gst/interfaces/xoverlay.c:
- Remove unused statement from doc example.
+2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-21 Stefan Kost <ensonic@users.sf.net>
-
- * gst-libs/gst/interfaces/videoorientation.c:
- (gst_video_orientation_iface_init),
- (gst_video_orientation_get_hflip),
- (gst_video_orientation_get_vflip),
- (gst_video_orientation_get_hcenter),
- (gst_video_orientation_get_vcenter),
- (gst_video_orientation_set_hflip),
- (gst_video_orientation_set_vflip),
- (gst_video_orientation_set_hcenter),
- (gst_video_orientation_set_vcenter):
- Add since tags to new API docs, ChangeLog surgery (forgot API keyword
- in ChangeLog)
+ * gst/playback/gstplaybin2.c:
+ Revert "playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler"
+ This reverts commit 7335ce5d3e03c126a417a721571cb6f3af136ecf.
+ Support abusing the uri property to configure the next uri to play
+ outside of the about-to-finish handler for the time being after all.
+ We also shouldn't use thread private structures for this, since it
+ should be possible to block the thread that emitted about-to-finish
+ while the main thread sets the uri property. See #607226.
-2006-09-21 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
- (create_rgb_conversions), (rgb_conversion_free),
- (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
- (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
- Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
- but disable for now since it doesn't pass (something wrong with
- RGBA somewhere).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't leak allocated buffers
+ This can happen if the combined flow return is not OK although the
+ allocation succeeded or if the packet in question is a BOS and we're
+ not going to push headers.
+ Fixes bug #608699.
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_commit),
- (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
- (queue_out_of_data), (gen_preroll_element),
- (preroll_remove_overrun), (probe_triggered):
- Refactor handling of overrun detection.
- Separate handling of group completion and deadlock detection when doing
- network buffering. This should fix some deadlocks that were not detected
- because the group was completed.
- Add more comments, improve debugging.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: clean up decodebin properties
+ When reusing a decodebin2 element, clear the properties we might have changed,
+ to their default values or else we might end up with old configuration.
+ Fixes #608484
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/libs/audio.c:
- Some more compilation fixes.
+ * gst/playback/gstplaybin2.c:
+ playbin2: when no uri is set, post an error message
+ When no uri is set, don't just return STATE_CHANGE_FAILURE from the
+ state change function, but actually post an error message.
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Early morning compilation fix.
+ * common:
+ Automatic update of common submodule
+ From 15d47a6 to 96dc793
-2006-09-20 Wim Taymans <wim@fluendo.com>
+2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/elements/multifdsink.c: (GST_START_TEST):
- * tests/check/elements/videorate.c: (GST_START_TEST):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
- Fix some warnings.
+ * gst/adder/gstadder.c:
+ adder: don't hold object lock when calling peer elements
+ Do not hold the object lock while we call methods on peer elements as this can
+ lead to deadlocks.
+ Fixes #608179
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
- Handcrafted merge to help CVS understanding what I changed and what
- not.
+ * configure.ac:
+ 0.10.25.2 pre-release
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_get_times):
- change colorkey behaviour back according to #354773 comment 6/7
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/video-enumtypes.c:
+ win32: update generated files for non-autotools win32 builds
-2006-09-19 Michael Smith <msmith@fluendo.com>
+2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
- (gst_multi_fd_sink_recover_client),
- (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
- (gst_multi_fd_sink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Implement stubbed out properties unit-type, units-soft-max,
- units-max, to allow specifying maximum sizes in units other than
- buffers.
- Fixes #355935
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translation files
-2006-09-19 Wim Taymans <wim@fluendo.com>
+2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Reorder the audio formats a bit for clarity.
- Detect and create caps for MSGSM and MSN (WAV49).
- Fixes #356596.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
- Small cleanups, move error handling out of normal flow for clarity.
+2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't skip an element when getting the topology
+ Fixes #608167
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs.types:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/videoorientation.c:
- (gst_video_orientation_get_type),
- (gst_video_orientation_iface_init),
- (gst_video_orientation_get_hflip),
- (gst_video_orientation_get_vflip),
- (gst_video_orientation_get_hcenter),
- (gst_video_orientation_get_vcenter),
- (gst_video_orientation_set_hflip),
- (gst_video_orientation_set_vflip),
- (gst_video_orientation_set_hcenter),
- (gst_video_orientation_set_vcenter):
- * gst-libs/gst/interfaces/videoorientation.h:
- API: Add new interface to control video orientation (fixes #354908)
+2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: sparse streams aren't timed by end time, and their duration isn't implicit
+ Fixes timestamps and durations on Kate subtitle streams.
+ See http://www.xiph.org/ogg/doc/ogg-multiplex.html section 'start-time and
+ end-time positioning' for some more details, and bug #600929.
- * gst/videotestsrc/gstvideotestsrc.c:
- Use G_UNLIKELY in _create and log one more detail.
-
- (gst_video_test_src_get_times), (gst_video_test_src_create):
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
- Use gst_util_uint64_scale_int in _get_times().
+2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: properly set up the media type for kate streams
+ See #600929.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
- Give better warning message (add object and detail).
+2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: relax caps template on sink pads
+ Allow any caps on sink pad templates as we could do passthrough with non raw
+ video caps.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_get_times):
- xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
- #354773), use gst_util_uint64_scale_int in _get_times()
+2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-18 Michael Smith <msmith@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: use right type for the serialno
+ Use a consistent type for the serialno to avoid problems when comparing between
+ signed and unsigned variants.
+ Fixes #607926
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
- Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
- always true, leading to dropping all timestamps.
+2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't push headers twice
+ Don't push the stream headers twice but only in the activation of a chain.
+ Fixes #607929
- * ext/libvisual/visual.c: (gst_vis_src_negotiate),
- (gst_visual_chain), (gst_visual_change_state):
- update to work also with libvisual 0.4 API, fix double unref (#355914)
-
- * tools/gst-launch-ext.1.in:
- * tools/gst-visualise.1.in:
- remove references to old man-pages
+2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (main):
- add real meadi-buttons, add tool-tips for the seek-options, arrange
- seek options in a table
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2006-09-18 Michael Smith <msmith@fluendo.com>
+2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
- (gst_ogg_mux_push_buffer):
- Don't generate out-of-order timestamps from oggmux, instead clamp
- output timestamps to be >= the previously output ts.
- Fixes #355595
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: rename a variable
+ Rename the 'seekable' variable to 'pullmode'. We might be able to seek in push
+ mode too eventually.
-2006-09-18 Michael Smith <msmith@fluendo.com>
+2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multi_fd_sink_class_init):
- Updates, fixes, and typo corrections for multifdsink. No functional
- changes.
+ * gst/playback/gstinputselector.c:
+ Revert "inputselector: Protect g_object_notify() with the object's mutex"
+ This reverts commit a37426c41c80fd21e5017fea01a786c05bcd9661, it's
+ causing deadlocks with playbin2.
-2006-09-17 Michael Smith <msmith@fluendo.com>
+2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
- * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
- Don't crash on truncated files - check that we got an 8 byte buffer
- before trying to memcmp it.
+ * gst/playback/gstinputselector.c:
+ inputselector: Protect g_object_notify() with the object's mutex
+ This works around the thread unsafety of g_object_notify()
+ Fixes bug #607513.
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (get_active_source):
- Make stream-switching appear instant to the application
- (ie. make sure that a g_object_get on 'current-foo' returns
- the stream previously set with g_object_set(). Totem needs
- this to update stream-related meta-info (like audio-codec)
- correctly when switching streams.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinder for ISO MP4 files
+ Fixes bug #607848.
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-24 13:29:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
- (gst_alsa_mixer_ensure_track_list):
- Try harder to guess which mixer track is the master mixer
- track (instead of just taking the first one that has a pvolume).
- Fixes #342228.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix crash when freeing headers
+ Use _ogg_packet_free() instead of gst_mini_object_unref in one more
+ place now that the header list contains ogg packets and not buffers.
+ file: Stephen_Fry-Happy_Birthday_GNU-nq_600px_425kbit.ogv
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- reviewed by: <delete if not using a buddy>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Strip trailing \0 for subtitle OGM streams
+ Fixes bug #607870.
- * gst-libs/gst/audio/audio.h:
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
+2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Correctly set DELTA_UNIT flag for OGM streams
- * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
- (gst_audio_convert_transform_caps):
- Get structure-name just once.
+2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't strip all 0-bytes from the end of OGM packets
+ This fixes broken packets pushed downstream by oggdemux for
+ MPEG4 streams for example.
- * tests/check/elements/audioresample.c: (GST_START_TEST):
- * tests/check/elements/videotestsrc.c: (check_rgb_buf):
- * tests/check/elements/volume.c: (GST_START_TEST):
- * tests/check/elements/vorbisdec.c: (GST_START_TEST):
- * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
- (test_pipeline), (GST_START_TEST):
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
- Fix big batch of compiler warnings.
+2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Extract tags from OGM text streams and don't push them downstream
- * ext/gnomevfs/gstgnomevfssrc.c:
- Add docs about icydemux usage in connection with gnomevfssrc
+2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/libvisual/visual.c:
- * ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- * gst-libs/gst/audio/gstaudiosink.c:
- * gst-libs/gst/audio/gstaudiosrc.c:
- * gst/audiorate/gstaudiorate.c:
- More G_OBJECT macro fixing.
-
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Fix wrong info in header due to copy & paste
+ oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
-2006-09-15 Wim Taymans <wim@fluendo.com>
+2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
- (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Do the delay calculation in the source/sink base classes as this is
- specific for the capture/playback mode.
- Try to fixate a bit better, like round depth up to a multiple of 8
- bigger than width.
- Handle underruns correctly by marking DISCONT on buffers and adjusting
- timestamps to handle the gap.
- Set offset/offset_end correctly on buffers.
-
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
- (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Remove resync and underrun recovery from the ringbuffer.
- Fix ringbuffer read code on under/overrun.
-
-2006-09-15 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (fill_buffer), (check_queue),
- (queue_threshold_reached), (gst_play_base_bin_set_property),
- (gst_play_base_bin_get_property):
- * gst/playback/gstplaybasebin.h:
- Don't use a 0 low watermark when buffering, it is catching starvation
- way too late. Instead, use a 3 second queue with 30 and 95
- percent low/high watermarks.
- Added queue-min-threshold property to configure low watermark.
- Use new _buffering message API.
- Make queue_threshold variable big enough to store a uint64 time value.
- API: playbin::queue-min-threshold property.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: optimise AC-3 typefinder a bit
+ Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
+ do gst_type_find_peek() in the inner loop all the time. Also return
+ when we've suggested AC3 caps, instead of continuing with the loop.
-2006-09-15 Wim Taymans <wim@fluendo.com>
+2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- We require 0.10.10.1 now because of _wait_preroll().
+ * gst/typefind/gsttypefindfunctions.c:
+ Revert "typefind: Reduce number of calls to gst_type_find_peek."
+ This reverts commit c661bfaa991c58f1fbd9fbc0dae90b8b2c27f92b.
+ This breaks AC-3 typefinding for all cases where the first frame
+ is at an offset > 0.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Use gst_base_sink_wait_preroll().
+2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-09-15 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for Zip Block Motion Video
- * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
- * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
- Use DEBUG_OBJECT more.
+2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
-=== release 0.10.10 ===
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Zip Block Motion Video
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
- patch by: Michael Smith <msmith at fluendo dot com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: YUNV is a fourcc which is also used for YUY2 raw video
- * gst/tcp/gstmultifdsink.c: (is_sync_frame),
- (gst_multi_fd_sink_client_queue_buffer),
- (gst_multi_fd_sink_new_client):
- * tests/check/elements/multifdsink.c: (GST_START_TEST),
- (multifdsink_suite):
- Fix implementation of sync-method 'next-keyframe'
- Closes #354594
+2010-01-23 15:13:45 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: vp61 and VP61 are also valid On2 VP6 fourcc
- patch by: Wim Taymans <wim at fluendo dot com>
+2010-01-23 15:10:45 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
- This patch removes the RANDOM flag that was incorrectly introduced with
- revision 1.91. Fixes #354590
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for On2 VP5
-2006-09-05 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 15:04:35 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/Makefile.am:
- Random variation in Makefile line to see if it makes the
- gen64-base-full bot any happier.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Sigma-Designs MPEG4
+ It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/pipelines/oggmux.c: (oggmux_suite):
- Disable test that fails at the moment (killed after timeout).
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for LOCO Lossless codec
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: James Livingston <doclivingston at gmail.com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for LOCO Lossless codec
- * tests/check/Makefile.am:
- * tests/check/pipelines/.cvsignore:
- * tests/check/pipelines/oggmux.c: (get_page_codec),
- (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
- (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
- (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
- (test_theora_vorbis), (oggmux_suite):
- Add simple unit test for oggmux from #337026 with checking for the
- EOS flags disabled for the time being.
+2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-09-04 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
- patch by: Alessandro Dessina <alessandro nnva org>
+2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/gstoggmux.c:
- Add cmml caps to oggmux. Fixes #353912
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Autodesk Animator codec
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/elements/videotestsrc.c: (check_rgb_buf):
- Returning a return value often helps. In this case, we
- don't need the return value anyway, so just get rid of it.
- Should make build bots much happier.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Autodesk Animator Codec
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
- (paint_get_structure), (gst_video_test_src_get_size),
- (gst_video_test_src_smpte), (gst_video_test_src_snow),
- (gst_video_test_src_unicolor), (paint_setup_AYUV),
- (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
- (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
- * gst/videotestsrc/videotestsrc.h:
- Add support for AYUV and the various RGBA formats. Initialise
- fields of paintinfo structs allocated on the stack.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: ...and set caps on queued packet buffers too
- * tests/check/elements/videotestsrc.c: (right_shift_colour),
- (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
- (check_rgb_buf), (videotestsrc_suite):
- Add unit tests for videotestsrc's RGB output.
+2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Set caps on header buffers
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_pattern_get_type),
- (gst_video_test_src_set_pattern):
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
- (gst_video_test_src_black), (gst_video_test_src_white),
- (gst_video_test_src_red), (gst_video_test_src_green),
- (gst_video_test_src_blue):
- * gst/videotestsrc/videotestsrc.h:
- Add more uni-colour patterns ("white", "red", "green", and "blue").
+2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: handle raw sources about-to-finish signals
+ When we are dealing with a source that produces raw audio/video, we don't use a
+ decodebin2 to decode the data and we thus don't have the drained/about-to-finish
+ signal emited. To fix this, we add a padprobe on the source pads and emit the
+ drained signal ourselves. This then makes playbin2 emit the about-to-finish
+ signal for raw sources such as cdda://
+ Fixes #607116
- * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
- Fix stride for YVYU, should be word-aligned (#353658).
+2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-08-31 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: include stdio.h for sscanf
- * gst/adder/gstadder.c: (gst_adder_src_event):
- Fix build.
+2010-01-22 01:49:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-31 Edward Hervey <edward@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add PNM typefinder
+ Add PNM typefinder, so we can remove the one that's in the PNM plugin
+ in -bad (which btw uses different/wrong media types that don't match
+ the ones used by gdkpixbufdec) and people don't make fun of us for
+ loading image decoders when typefinding and playing back audio files.
- * gst/adder/gstadder.c: (forward_event_func),
- (gst_adder_src_event), (gst_adder_collected),
- (gst_adder_change_state):
- * gst/adder/gstadder.h:
- Remember the start position asked in the incoming seeks, so we can
- output GST_EVENT_NEW_SEGMENT with a correct position value (instead
- of assuming it will always be 0).
+2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
-2006-08-31 Edward Hervey <edward@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: rename performance category
+ rename the performance category to ffmpegcolorspace_performance
+ as there is already a global GST_CAT_PERFORMANCE in core
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
- (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_loop):
- Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
+2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-30 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: keep track of added pads
+ Keep track of the pads we added and removed.
+ Remove some unused fields.
+ Don't add pads for which we don't have caps.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size):
- Return FALSE instead of returning a random false unit
- size when the format isn't known/supported (even if
- this shouldn't happen under normal circumstances).
+2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-29 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggstream.c:
+ oggstream: don't call NULL setup functions
+ If we find a known mapper but it doesn't have a setup function, simply skip it
+ instead of crashing.
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
+2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
- (gst_gnome_vfs_src_start):
- Try harder to get the size from a uri by using _info_uri() when
- _info_from_handle() does not give us enough info.
- Also follow symlinks when getting the size.
- Partially Fixes #332864.
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoid division by 0 on bad annodex streams
-2006-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: Viktor Peters <viktor dot peters at gmail dot com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for y4m container
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
- (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
- (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
- (gst_alsa_mixer_set_record):
- * ext/alsa/gstalsamixertrack.c:
- (gst_alsa_mixer_track_update_alsa_capabilities),
- (alsa_track_has_cap), (gst_alsa_mixer_track_new),
- (gst_alsa_mixer_track_update):
- * ext/alsa/gstalsamixertrack.h:
- Improve and fix mixer track handling, in particular better handling
- of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create
- separate track objects for tracks that have both capture and playback
- volume (and label them differently as well so they're not mistakenly
- assumed to be duplicates); classify mixer tracks that only affect
- the audible volume of something (rather than the capture volume)
- as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
- for capture tracks to correspond to alsa-pswitch alsa-cswitch
- (following the meaning documented in the mixer interface header
- file); add support for alsa's exclusive cswitch groups; update/sync
- state/flags better if mixer settings are changed by another
- application. Fixes #336075.
+2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-08-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: ptime/maxptime should be unsigned
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
- * gst/playback/gstplaybin.c:
- Improve docs: add section about BUFFERING messages sent by playbin.
+2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-08-29 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: ptime should be in nanoseconds
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
- (gst_vorbis_enc_buffer_check_discontinuous),
- (gst_vorbis_enc_chain):
- Ignore explicit DISCONT marked on buffers (which is often spurious,
- particularly when using multiple segments), in favour of solely
- using the timestamps/durations.
+2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-29 Edward Hervey <edward@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 14cec89 to 15d47a6
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Don't rely on incoming buffers offset anymore, since it is completely
- broken when using multiple segments.
- Instead convert the incoming buffers timestamp to running time, and
- then convert that value to the offsets.
- Also inform GstSegment of the last outputted stop position, which is
- needed if we received several segments with an unknown stop value.
+2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
-2006-08-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: rewrite h.264 detection
+ Make detection simpler: check for NALs, check that they make
+ sense, and report how certain we are that it's a raw H.264 stream.
+ Fixes: #583376.
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- fix buffer unreffing on a header push failure
+2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-08-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Reject empty caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=607353
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
- (gst_audio_rate_chain):
- Make the metadata of the buffer writable before changing its
- flags.
+2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2006-08-28 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: No need to subtract begin time
+ Last stop is already based on the chain start and there is no need
+ to subtract the chain start as it may lead to a negative overflow.
+ This was causing seeking issues when the target chain was not
+ the first one (that has chain start = 0)
+ Fixes #606382
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
- (gst_audio_rate_setcaps), (gst_audio_rate_init),
- (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
- (gst_audio_rate_chain), (gst_audio_rate_change_state):
- Fix audiorate some more.
- Reset and resync counters on flush and READY.
- Handle the DISCONT flag correctly.
- Use GstSegment to track position.
- Fail when not negotiated.
- Fixes #353234.
+2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-25 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/audio.h:
+ audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
+ Fixes bug #607381.
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Fix spelling.
- Remove accidently included debug line.
+2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2006-08-25 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: granulepos is relative to its chain
+ When performing seeks, the granulepos should be offset by
+ its chain start time to avoid using wrong values to
+ update segment's last_stop. A sample file is indicated on
+ bug #606382
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Small cleanups.
- If a buffer is received with no caps, make the buffer metadata
- writable and set the caps, making sure that we don't screw up the
- refcounts.
+2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-08-25 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for MXF container format
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
- (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
- Fix memory leaks and misleading debug messages, add a couple of
- comments.
+2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
- (gst_multi_fd_sink_render):
- Do not use gst_buffer_make_writable() in a basesink render method,
- as it may incorrectly unref the buffer. Instead, use convoluted
- dance to avoid copying the buffer except when we need to.
+ * gst/playback/gstplaysink.c:
+ playsink: re-use iterator callback to avoid code duplication
-2006-08-25 Michael Smith <msmith@fluendo.com>
+2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisenc.c:
- (gst_vorbis_enc_buffer_check_discontinuous):
- Allow very small discontinuities in the timestamps. These we can't
- do anything useful with anyway (because vorbis's timestamps have
- only sample granularity), and are commonly produced by elements with
- minor bugs. Allow up to 1/2 a sample out.
- Fixes #351742.
+ * gst/playback/gstplaysink.c:
+ playsink: when looking for sink properties, make sure they have the right type
+ We don't want to end up setting values on elements where the property is of
+ a different type than we expect. Can't transform the value either, since we
+ can't really make assumptions about the scale and transform function.
+ Fixes crashes when using playbin2 with apexsink (#606949).
-2006-08-24 Wim Taymans <wim@fluendo.com>
+2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
- (play_scrub_toggle_cb), (main):
- Add a checkbox to enable play scrubbing. Makes it possible to disable
- normal scrubbing.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler
+ Changing the URIs in a state > READY results in unexpected behaviour,
+ i.e. the new URIs are only used after the current track has finished.
+ Fixes bug #607226.
-2006-08-23 Stefan Kost <ensonic@users.sf.net>
+2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/elements/.cvsignore:
- make buildbot happy
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: sprinkle some more locking
+ ... to avoid races and ensure some data structure consistency.
+ See also #574289.
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
- (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
- (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
- (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
- (gst_ogm_text_parse_strip_trailing_zeroes),
- (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
- (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
- Refactor ogm parse, do better input checking, misc. clean-ups.
- Cache incoming events and push them once the source pad has
- been created. Don't pass unterminated strings to sscanf().
- Strip trailing zeroes from subtitle text output, since they
- are not valid UTF-8. Don't push vorbiscomment packets on
- the subtitle text pad. Output perfect streams if possible.
+2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-23 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: mind blocked pads when shutting down
+ Fix regression in shutdown deadlock handling now that the
+ target of a ghostpad is blocked instead of ghostpad itself.
+ See also #574293.
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- Waits for tasks to settle down so that we clean up correctly for
- valgrind.
+2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Fix disabling of subtitles if subtitles were used before
+ In this case the video still goes through the text chain and
+ subtitles are still going in there, in case subtitles are
+ enabled again. This makes sure that re-enabling subtitles
+ happens instantly.
+ Fixes hanging video when disabling subtitles, caused by an
+ unliked video pad.
- * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
- Unit test fixes: \377 is more likely to fit into 8 bits than \777;
- actually return return value in taglists_are_equal.
+2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix pad ref leak
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Fix crash due to broken bitstream parsing on x86-64: can't make
- any assumptions about sizeof(struct) due to alignment/packing
- differences on different architectures. Fixes #351790.
+2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-22 Wim Taymans <wim@fluendo.com>
+ * docs/plugins/Makefile.am:
+ docs: fix out-of-source build
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
- (gst_riff_parse_chunk), (gst_riff_parse_file_header),
- (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
- (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
- (gst_riff_parse_info):
- Protect public functions against bad input.
- Do some cleanups.
- Fix documentation.
+2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/stress-playbin.c:
+ stress-playbin: fix error return check
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Add voxware audio IDs (even if we can't play it) (#351795).
+2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/Makefile.am:
+ * ext/theora/gsttheora.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/theora/theora.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c:
+ theora: Rename source files to have the same name as the headers
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps),
- (gst_riff_create_iavs_template_caps):
- Const-ify some arrays and use G_N_ELEMENTS instead
- of wasting oodles of RAM on terminator bits.
+2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/gstvorbis.c:
+ * ext/vorbis/gstvorbisdec.c:
+ * ext/vorbis/gstvorbisenc.c:
+ * ext/vorbis/gstvorbisparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * ext/vorbis/vorbis.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbistag.c:
+ vorbis: Rename source files to have the same name as the headers
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_to_vorbiscomment_buffer):
- * tests/check/libs/tag.c: (GST_START_TEST):
- And the same for _to_vorbiscomment_buffer(): allow
- id_data_len == 0 for speex.
+2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/gstvorbiscommon.c:
+ * ext/vorbis/gstvorbiscommon.h:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: Move channel layout definitions into a single separate file
+ ...instead of having two copies.
- * configure.ac:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-gdp.xml:
- * gst/gdp/Makefile.am:
- * tests/check/Makefile.am:
- Move GDP plugin to -base from -bad. Closes #347783.
+2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: Add official 6.1 and 7.1 channel mappings
+ These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_from_vorbiscomment_buffer):
- Allow id_data_len == 0 (needed for vorbis comments in Speex files).
- Also add some checks to make sure we don't memcmp() beyond the end of
- vorbiscomment buffer if the ID to check for is larger than the buffer.
+2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/libs/tag.c: (GST_START_TEST):
- Some more tests for gst_tag_list_from_vorbiscomment_buffer().
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Don't define h_error ourselves
+ It's included from netdb.h and that header might define it differently,
+ which can lead to build failures.
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
- (gst_vorbis_enc_set_metadata):
- Use vorbis comment utility functions from libgsttag
- instead of re-inventing the wheel (partially fixes #347091).
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: mp4 video is not parsed
-2006-08-21 Jan Schmidt <thaytan@mad.scientist.com>
+2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Fix leaks. Wait for state transitions that might happen ASYNC, as well
- as some that won't.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add aac stream-format to caps
+ Also add the aac stream-format field on the caps when
+ detecting it.
-2006-08-21 Wim Taymans <wim@fluendo.com>
+2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- Don't try to GObject scan the netbuffer as it's not a GObject.
- Fixes #351308.
+ * gst/playback/gstplaysink.c:
+ playsink: Fix handling of the native audio/video flags
+ Fixes bug #606687.
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- Document GstNetBuffer.
+2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-08-21 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix unitialized variable.
+ If the package isn't handled, gracefully return GST_FLOW_OK.
- * tests/check/elements/audioconvert.c: (GST_START_TEST),
- (audioconvert_suite):
- Add testcase for caps-size-explosion
+2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-20 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/interfaces/xoverlay.c:
+ docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
+ Explain why the whole bus sync handler mess is needed. Add section about
+ how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
+ break issue and how to work around it (see #601809).
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_unit_size), (set_structure_widths):
- Lower debug, use g_assert in _get_unit_size
+2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c:
- (audioresample_get_unit_size):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
- use g_assert in _get_unit_size
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ docs: minor netbuffer documentation fix
-2006-08-18 Wim Taymans <wim@fluendo.com>
+2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
- (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
- (gst_rtp_buffer_get_payload_buffer):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Document GstRTPBuffer.
- Added function to efficiently strip payload headers.
- API: gst_rtp_buffer_get_payload_subbuffer()
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translated strings
+ Queue2 moved into core, so remove its strings.
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
- (gst_tag_to_vorbis_comments):
- Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
- tags and deserialise them properly as well (#347091).
- Add some more gtk-doc blurbs and also some g_return_if_fail().
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: push headers when activating chains
+ Keep a list of headers for each stream of a chain. When a chain is activated,
+ push the headers before pushing the data so that decoders can sync.
+ Fix seeking in chains, take the chain start time into account when comparing
+ timestamps.
+ See #606382
- * tests/check/libs/tag.c: (GST_START_TEST),
- (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
- More tests.
+2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-17 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/lang-tables.c:
+ * gst-libs/gst/tag/lang-tables.dat:
+ * gst-libs/gst/tag/lang.c:
+ tag: fix up disting of lang-tables.c more correctly
+ lang-tables.c is included by lang.c and not really a proper source
+ file that should be compiled into its own object, so rename it to
+ lang-tables.dat and put it into EXTRA_DIST instead to ensure it
+ gets disted.
- * ext/ogg/Makefile.am:
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
- (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
- (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
- (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
- (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
- (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
- Added ogg-in-avi parser element. Fixes #140139.
+2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
- Fixed a bug in oggdemux debug code.
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-plugins-base.spec.in:
+ Add missing source file for tagger to Makefile and update spec file
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Recognise Ogg in the AVI extensible wave format.
+2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: handle 32 bit raw RGB video.
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
- Make buffer durations add up (duration should be next_ts-ts for
- perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
- from CVS.
+2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
- (test_buffer_timestamps), (cddabasesrc_suite):
- Add unit test for the above.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: decide flac header packet by content rather than count
- * tests/check/Makefile.am:
- Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
- to see what happens.
+2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-16 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: reset header packet count at bos page
- * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
- (gst_alsasink_open):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
- (gst_alsasrc_open):
- Avoid setting and using a NULL device name.
- Print more info when we fail to open a device.
+2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: add support for buffer-lists
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
- API: add gst_tag_parse_extended_comment() (#351426).
+2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/libs/.cvsignore:
- * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
- Add unit test for gst_tag_parse_extended_comment().
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
- Fix leak (#351502).
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Ignore zero framerate
+ https://bugzilla.gnome.org/show_bug.cgi?id=606163
-2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * gst/playback/gstplaybin.c:
- Document playbin.
-
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-cdparanoia.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- * docs/plugins/inspect/plugin-playbin.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml:
- Update to CVS version.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ basertpaudiopayload: Respect ptime if it is given
+ If the ptime is given in the caps, respect it and force the minimum
+ and maximum sizes to be exactly the requested ptime.
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (value_list_append_structure_list),
- (gst_play_bin_handle_redirect_message),
- (gst_play_bin_handle_message):
- API: GstPlayBin::connection-speed
- Add "connection-speed" property; re-order redirect messages with
- multiple redirect locations depending on the minimum bitrate if
- that information is available and a connection speed is set
- (#350399).
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ rtpbasepayload: Store ptime from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Update max volume to the same value that the volume element uses.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Accept maxptime from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
-2006-08-14 Wim Taymans <wim@fluendo.com>
+2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Less uglyness..
+ * ext/ogg/gstoggstream.c:
+ oggdemux: enhance flac packet duration calculation
-2006-08-14 Wim Taymans <wim@fluendo.com>
+2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
- (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
- Add some more debug info.
- Don't crash when a seek failed.
- Actually return the result of the seek instead of TRUE.
- Ignore multiple BOS pages with the same serial so that we don't create
- the same stream multiple times.
- Post an error when we fail to do the initial seek.
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2006-08-13 Wim Taymans <wim@fluendo.com>
+2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
- (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
- Small code cleanup.
+ * tests/examples/seek/seek.c:
+ * tests/icles/test-colorkey.c:
+ examples: use Gtk+-2.18 API conditionally
+ so the seek example and colorkey test work with older Gtk+ versions
+ as well.
+ Fixes #605960.
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
- (gst_alsa_mixer_new):
- Remove hack that always set the device to hw:0*.
- Properly find the card name for whatever device was configured.
- Do some better debugging.
- Fixes #350784.
+2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_set_property),
- (gst_alsa_mixer_element_change_state):
- Cleanups.
- Handle setting of a NULL device name better.
+ * tests/icles/test-colorkey.c:
+ tests: fix colorkey test up for Gtk+ >= 2.18
+ Make test-colorkey work with newer versions of Gtk+.
+ See #601809.
-2006-08-11 Wim Taymans <wim@fluendo.com>
+2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c:
- Don't clip float values. Fixes #350900.
+ * tests/examples/seek/seek.c:
+ examples: make seek example work with Gtk+ >= 2.18
+ Gtk+ broke API slightly with the introduction of
+ client-side windows in Gtk+ 2.18. Fix up seek
+ example to work with newer Gtk+ versions.
+ Fixes #601809.
-2006-08-11 Andy Wingo <wingo@pobox.com>
+2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gsttcp.c: Really fix the build?
+ * tests/icles/stress-xoverlay.c:
+ tests: fix warning and memory leak in stress-overlay test
+ Not all messages have structures and we need to unref messages
+ when returning GST_BUS_DROP in the sync bus handler.
- * gst/tcp/gsttcp.h: For now, always disable deprecation here --
- fixes the build.
+2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: correctly eat empty and dummy buffers
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
- Float caps shouldn't have a "signed" field.
+2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ adder: be a lot smarter with buffer management
+ Detect EOS faster.
+ Try to reuse one of the input buffer as the output buffer. This usually works
+ and avoids an allocation and a memcpy.
+ Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
+ try to use a GAP buffer as the output buffer when all input buffers are GAP
+ buffers.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
- Implement SEEKING query in its most basic form, so that we can
- at least check if we're seekable or not (#350655).
+2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/Makefile.am:
+ * gst/adder/gstadder.c:
+ * tests/check/elements/adder.c:
+ adder: use collectpads clipping function
+ Install a clipping function in the collectpads and use the audio clipping helper
+ function to perform clipping to the segment boundaries.
+ Fixes #590265
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- The checks here are not even close to anything that would
- justify MAXIMUM probability, lowering to POSSIBLE until someone
- fixes the checks (case at hand: quicktime redirection files
- might start with 00 00 01 XX and pass the checks here just
- fine, see #350399).
+2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ adder: fix juvenile comment
- Patch by: Sjoerd Simons <sjoerd at luon net>
+2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
- Better detection for multipart/x-mixed-replace: accept leading
- whitespaces before the boundary marker as well (as our very own
- multipartmux used to produce) (#349068).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix typo in debug message
-2006-08-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some type checks
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Detect DTS audio streams (#350157).
+2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-05 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: avoid leaking selector request pads
- * ext/theora/gsttheoraparse.h:
- * ext/theora/theoraparse.c (gst_theora_parse_class_init)
- (theora_parse_dispose, theora_parse_set_property)
- (theora_parse_get_property, theora_parse_munge_granulepos)
- (theora_parse_push_buffer, theora_parse_change_state):
- API: GstTheoraParse::synchronization-points
- Add a property 'synchronization-points' to fix badly synchronized oggs.
+2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: avoid leaking queue and typefind
+ Don't leak the queue and typefind elements that we might link after the
+ source element.
- * tests/check/Makefile.am:
- * tests/check/libs/.cvsignore:
- * tests/check/libs/audio.c: (structure_contains_channel_positions),
- (fixed_caps_have_channel_positions), (GST_START_TEST),
- (audio_suite), (main):
- Add a few tests for the channel position stuff in libgstaudio.
+2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: don't name the queue
+ There is no reason to name the queue.
+ Fixes #605219
- * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
- (gst_alsa_detect_channels):
- * ext/alsa/gstalsasink.c:
- Add support for cards that (only) do more than 8 channels,
- like the Delta 44 (#345188).
+2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions):
- * gst-libs/gst/audio/multichannel.h:
- API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
- unspecified channel position and cannot be combined with any
- of the other audio channel positions; adjust position layout
- checks accordingly (#345188).
+ * win32/common/libgstrtp.def:
+ defs: update defs with new symbols
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Recognise ancient RealAudio files (see #349779).
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ rtcpbuffer: add helper functions for SDES types
+ Add functions to convert SDES names to their types and back. Will be used later
+ to set SDES items using a GstStructure.
+ See #595265
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Patch by: Jens Granseuer <jensgr at gmx net>
+ * common:
+ Automatic update of common submodule
+ From 47cb23a to 14cec89
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for Interplay's MVE format (#348973).
+2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-08-02 Wim Taymans <wim@fluendo.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: add Since marker for the new tolerance property
- Patch by: Marcel Moreaux <marcelm at luon dot net>
+2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Handle RTP sequence number rollover.
- Disable jitterbuffer by default.
+ * gst-libs/gst/tag/lang.c:
+ docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
-2006-07-28 Jan Schmidt <thaytan@mad.scientist.com>
+2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c: (audioresample_stop),
- (audioresample_set_caps):
- Don't leak references to the incoming caps. Clean them up when
- stopping.
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ * tests/examples/app/appsrc-stream.c:
+ * tests/examples/app/appsrc-stream2.c:
+ tests: don't use deprecated GLib API g_mapped_file_free
+ Fixes #605100.
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
- (gst_video_scale_finalize):
- Don't leak our temporary pixel buffer.
+2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
- * tests/check/Makefile.am:
- * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
- (GST_START_TEST), (simple_launch_lines_suite):
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theoraenc: Add encoder controls for libtheora 1.1
+ Added drop-frames, cap-overflow, cap-underflow, and rate-buffer.
- Fix leaks and re-enable the test for valgrind checking.
+2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: increase default drift tolerance to fix glitches with WMA
+ Increase default drift tolerance to 40ms to avoid glitches with decoders
+ or formats where there's a lot of timestamp jitter for some reason or
+ another (in this case: asf/wma), at least until we implement timestamp
+ smoothing.
- Patch by: Sjoerd Simons <sjoerd at luon net>
+2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
- (plugin_init):
- Add typefind function for multipart/x-mixed-replace (#348916).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add some debugging
-2006-07-28 Wim Taymans <wim@fluendo.com>
+2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_query_duration):
- Fix leak in duration query.
- Reflow some docs and notes.
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ audiorate: add a tolerance property
+ It may not be uncommon for the input timestamps to experience some jitter
+ around the 'perfect time'. As such, instead of regularly adding and dropping
+ samples, optionally allow for some tolerance in a more relaxed approach.
+ API: GstAudioRate:tolerance
-2006-07-28 Michael Smith <msmith@fluendo.com>
+2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
- (vorbisenc_suite):
- Enable Andy's extra vorbisenc test, now that it passes. Also fix one
- aspect of it.
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: add documentation
-2006-07-28 Michael Smith <msmith@fluendo.com>
+2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
- (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
- (gst_vorbis_enc_push_buffer),
- (gst_vorbis_enc_buffer_check_discontinuous),
- (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
- * ext/vorbis/vorbisenc.h:
- Handle discontinuities in the input vorbis stream correctly,
- so that the output is properly timestamped (and has good granulepos
- values). Needs some oggmux fixes too.
+ * gst/audiorate/Makefile.am:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ audiorate: use separate header file
-2006-07-27 Wim Taymans <wim@fluendo.com>
+2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- patch by: Kai Vehmanen <kv2004 eca cx>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: set DISCONT when resyncing (e.g. newsegment)
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_handle_sink_event),
- (gst_base_rtp_depayload_change_state):
- Don't send multiple newsegments with different formats.
- Fixes #348677.
+2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-07-26 Wim Taymans <wim@fluendo.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: also fill up segments if possible
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
- (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
- Make seeking in ogg more accurate again by doing the more correct
- granuletime to stream time conversion.
+2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: fix segment handling
+ Do not compare a media (buffer) time to a (bogus) running time
+ (or their offset equivalents).
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
- (gst_multi_fd_sink_new_client):
- debug a little more understandably
- do not use goto as a substitute for break, especially if
- break is also being used
+2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: properly report truncated samples as dropped samples
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Remove GLib-2.6 compatibility cruft.
+2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-24 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/tag/lang.c:
+ docs: mention that gst_tag_get_language_name() may return NULL
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Don't try to align a sample to an unknown value.
+2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-24 Wim Taymans <wim@fluendo.com>
+ * tests/check/libs/tag.c:
+ checks: some more testing for the new language code functions
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
- When the audio clock is slaved to another clock, never try to align
- samples but trust the rate interpolation algorithm.
+2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-24 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ docs: misc. mixer docs improvements
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- Don't try to calculate silence samples, base class does this much
- better now.
+2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
- (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
- (gst_ring_buffer_acquire):
- Calculate silence samples correctly.
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ docs: add short descriptions for API reference contents page
- * gst-libs/gst/audio/gstringbuffer.h:
- Add _CAST macro.
+2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/lang-tables.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ tag: make internal language names table static
- * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
- Limit search for the first markup tag to the first few kB of
- the file. If we don't find one there, it's highly unlikely that
- this is an XML(-ish) file.
+2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-21 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/tag/lang.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ tag: don't use GLib 2.22 API
+ g_mapped_file_unref() was introduced in GLib 2.22, but we depend
+ only on GLib 2.18, so use g_mapped_file_free() when compiling
+ against older GLib versions until we bump the GLib dependency.
- * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
- test to the one in vorbisenc. Also commented out.
+2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/pipelines/vorbisenc.c:
- (test_discontinuity): New test, commented out until Mike lands
- some elite vorbisenc patches.
+ * .gitignore:
+ * configure.ac:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/lang-tables.c:
+ * gst-libs/gst/tag/lang.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ * gst-libs/gst/tag/tag.h:
+ * tests/check/libs/tag.c:
+ * win32/common/libgsttag.def:
+ tag: add some utility functions for language codes and tags
+ Add some utility functions for language tags and ISO-639
+ codes. These are useful for both GUIs and elements. The
+ iso-codes package is used for language name translations
+ if available.
+ API: gst_tag_get_language_codes()
+ API: gst_tag_get_language_name()
+ API: gst_tag_get_language_code()
+ API: gst_tag_get_language_code_iso_639_1()
+ API: gst_tag_get_language_code_iso_639_2B()
+ API: gst_tag_get_language_code_iso_639_2T()
- * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
- Bufferstraw was actually factored out of these tests. Now we share
- code yay.
+2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
- for bufferstraw addition to gstcheck.
+ * ext/ogg/gstoggstream.c:
+ ogg: ogm video has constant packet duration
-2006-07-21 Wim Taymans <wim@fluendo.com>
+2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
- * ext/theora/theoradec.c: (clip_buffer):
- Better clipping.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: implement old fLaC mapping
-2006-07-21 Wim Taymans <wim@fluendo.com>
+2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
- (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
- Fix leak.
- Avoid type casting when we can.
+ * gst/tcp/gsttcpclientsrc.c:
+ tcpclientsrc: unset flushing state too
+ When unlocking, we set the flushing state on the fdset. Implement unlock_stop so
+ that we can use it to unset the flushing state again.
+ Fixes #577326
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
- Fix mem leak.
+2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-20 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: remove redundant fields
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_change_state):
- Make state change fail if the specified device can't be opened
- for some reason.
+2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-20 Wim Taymans <wim@fluendo.com>
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: adapt to new oggdemux
+ Remove all granulepos hacks and simply use the timestamps from the new oggdemux
+ like any other decoder.
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (cb_newpad), (main):
- Example of a small audio/video player using decodebin.
+2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-20 Stefan Kost <ensonic@users.sf.net>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: fix peer query
- * gst-libs/gst/riff/riff-ids.h:
- Add 'fact' chunk id
+2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-19 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoradec.c:
+ theoradec: fix query
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_change_state):
- Don't assert when not negotiated but post a meaningfull
- error message. Fixes #347918.
+2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Add comment about better default MTU size.
+ * ext/theora/theoradec.c:
+ theoradec: small cleanups
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- Small cleanups, start docs.
+2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: use gst_pad_peer_query()
- Patch by: Martin Szulecki
+2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
- If "device-name" is requested and the device is not
- open, try to temporarily open it to obtain this
- information (#342494).
+ * gst/playback/gstplaysink.c:
+ playsink: fix video when subtitles disabled
+ When we have a source with subtitles but they were disabled with the flags,
+ still ghostpad the video pad instead of leaving it unlinked.
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c:
- Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only flush downstream on seeks for flushing seeks
- * gst-libs/gst/tag/gsttageditingprivate.h:
- * gst-libs/gst/tag/gstvorbistag.c:
- Some more random const-ifications.
+2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-18 Stefan Kost <ensonic@users.sf.net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_template_caps):
- Add more FOURCCs (sort list to make stuff easier to find),
- add comment what those 16 bytes in struct _gst_riff_strh according to
- one avi-dumper are
+2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: update slider only 25 times a second
+ don't update the slider a 100 times a second, it's likely higher than the screen
+ framerate and just wastes cpu.
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_fixate_channel_positions):
- Const-ify two arrays.
+2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: remove granulepos hacks
+ Remove the granulepos hacking now that oggdemux outputs timestamps like any
+ other demuxer.
- * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
- Fix typo, so that alsasink also advertises 8 channels
- if that's supported (tags: can, worms, open, alsa, ph34r).
+2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-17 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix stream-changed message list iteration
+ When iterating the list and removing the current element, first
+ get the next element and then remove the current one and not
+ the other way around.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
- (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
- *sigh*, when is the compiler going to warn when the comments
- are out-of-sync with the code.. Refix case of busted theora
- headers with 0 granule pos.
+2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-14 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: improve keyframe seeking
+ Improve keyframe seeking.
+ Fix reverse playback.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_wait),
- (gst_base_rtp_depayload_change_state),
- (gst_base_rtp_depayload_set_property),
- (gst_base_rtp_depayload_get_property):
- Fix 99% cpu load by waiting for absolute times on the
- clock. Fixes #347300.
-
-2006-07-14 Andy Wingo <wingo@pobox.com>
-
- * ext/theora/gsttheoraparse.h:
- * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
- (theora_parse_push_headers, theora_parse_clear_queue)
- (theora_parse_drain_queue_prematurely, )
- (theora_parse_sink_event, theora_parse_change_state): Queue events
- until we initialized our state, like in vorbisparse.
-
- * ext/vorbis/vorbisparse.h:
- * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
- (vorbis_parse_push_headers, vorbis_parse_clear_queue)
- (vorbis_parse_drain_queue_prematurely, )
- (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
- until we have initialized our state. Fixes seeking after an
- initial pad block.
-
-2006-07-14 Andy Wingo <wingo@pobox.com>
-
- Patch by: Iain Holmes <iaingnome@gmail.com>
-
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
-
-2006-07-14 Jan Schmidt <thaytan@mad.scientist.com>
+2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Bump nano back to CVS
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: implement keyframe seeking
+ Implement keyframe seeking in oggdemux by doing the double seek trick. First
+ seek to the required position, then read pages for all streams to grab the
+ granulepos (to know the timing of the keyframe) of each stream, then seek back
+ to the first keyframe.
-=== release 0.10.9 ===
+2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstplaysink.c:
+ playsink: Some minor cleanup
- * configure.ac:
- releasing 0.10.9, "I walk the line"
+2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-14 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
- * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
- Move a g_cond_signal to earlier to avoid sometimes deadlocking
- (commonly happens when running this test under valgrind) when trying
- to remove the buffer probe.
+2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix timestamps after seek
+ After a seek, discard all packets before the packet with the granulepos on it so
+ that the output buffers contain valid timestamps.
+ Reorder some code so that we check the timestamps before allocating and pushing
+ an output buffer.
+ Do more checks on valid packets in ogm mode.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Fix missing g_unlock from the previous commit
+2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add comment
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_change_state):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_change_state):
- Implement a locking order to ensure we always take the object lock
- before the x_lock and never vice-versa.
+2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't do math with invalid granulepos
+ When the current granulepos is unknown and set to -1, don't try to add durations
+ to it.
- * gst/playback/gstdecodebin.c: (find_compatibles):
- Fix a caps leak when linking (#347304)
+2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_change_state):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
- Don't leak shared memory resources. Use the object lock to protect
- against the xcontext disappearing while returning a buffer from the
- pipeline. (#347304)
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: guard against wrong granulepos
+ Clamp the initial granulepos to 0 instead of going negative for some badly muxed
+ ogg files.
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
- (vorbis_handle_comment_packet):
- gst_tag_list_merge() returns a new object. Take that into account when
- using it. This avoids memleak.
- Revert previous commit which is not needed.
+ * ext/theora/theoradec.c:
+ theoradec: don't fail on bogus granulepos
+ Do some additional checks on the granulpos timestamp before using it for
+ calculating the duration because oggdemux generates wrong granulepos now.
+ Fixes seeking somewhat again.
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
- Reset the decoder in finalize so that all fields get cleared.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: reimplement OGM support
+ OGM demuxing no longer requires helper elements. It's done internally
+ in oggdemux. Vorbis comments are still not handled because I don't
+ have anything to test with.
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_set_clock),
- (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
- Don't try to post an error message when setting the clock fails
- as this can happen when adding an element to a bin which will then
- deadlock. Fixes #347296.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: fix for I-frame-only theora
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-12-03 01:16:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
- (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
- (vorbis_handle_type_packet):
- Post tag messages on the bus even if we're not initialized.
- If we're not initialized, we still postpone the event pushing of tags.
+ * ext/ogg/gstoggstream.c:
+ ogg: log when ogg mapper doesn't accept the setup header packet
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
- (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
- Revert last two changes that broke the freeze.
+ * ext/ogg/gstoggstream.c:
+ ogg: extract width, height and PAR from theora header and add to caps
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- basesink calculates silence sample correctly for us.
+ * ext/ogg/gstoggstream.c:
+ ogg: extract number of channels from FLAC, speex and vorbis headers
+ Because we can.
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
- (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
- Calculate correct silence samples so we don't fill our ringbuffer
- with noise.
+ * gst/playback/gstplaybin2.c:
+ build: fix build with debug logging disabled.
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
- (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
- * ext/vorbis/vorbisdec.h:
- Delay sending events (newsegment, tags) until the decoder is properly
- initialized.
- Fixes #347295
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: more print fixes
+ gstoggstream.c:419: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘gint64’
+ gstoggdemux.c:2253: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
+ gstoggdemux.c:2333: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
-2006-07-11 Jan Schmidt <thaytan@mad.scientist.com>
+2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (get_float_mc_caps),
- (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
- Patch from #347221 adding a test for audioconvert
- channel remappings.
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Fixing some printf format strings
+ Fixes some printf format strings to make it build on mac.
-2006-07-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
- (gst_ssa_parse_parse_line):
- Don't include the terminating NUL in the buffer size,
- it's only there for extra paranoia (would add random
- '*' characters at the end of each subtitle since the
- terminator itself is not valid UTF-8 technically).
- Also fix indenting after boilerplate macro.
+ * gst/playback/gstfactorylists.c:
+ * gst/playback/gstfactorylists.h:
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't iterate the factory lists in non-debug mode
+ When debugging is disabled, we won't see anything printed anyway.
-2006-07-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Also emit 'unknown-type' signal (which should really be
- called unhandled-type) if we found potential decoders/demuxers
- in the registry but none of them worked in the end (as in the
- case where the plugins don't exist any longer but are still
- listed in the registry). Fixes #329798.
+ * gst/videoscale/vs_4tap.c:
+ Build fix for MSVC
-2006-07-08 Andy Wingo <wingo@pobox.com>
+2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
- * theoraparse.c (theora_parse_push_buffer)
- (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
- Add some more debugging. Fix granulepos reconstruction in the face
- of discontinuities.
+ * gst/subparse/qttextparse.c:
+ build: add missing includes for sprintf and atoi
-2006-07-06 Wim Taymans <wim@fluendo.com>
+2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init),
- (gst_base_audio_sink_provide_clock):
- Use gobject_class instead of G_OBJECT_CLASS (klass)
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/qttextparse.c:
+ subparse: Add support for some tags of qttext
+ Currently supporting timescale, timestamps, font, size,
+ textColor, backColor, plain, bold and italic
+ Fixes #603357
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init), (gst_base_audio_src_init),
- (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
- (gst_base_audio_src_get_time),
- (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
- (gst_base_audio_src_create_ringbuffer):
- Fix latency and buffer-time constants and properties ala basesink.
- Implement pull based scheduling. Fixes #346527.
- Set default blocksize in GstBaseSrc to 0, we default to pushing out
- one segment.
- Refuse slaving to another clock instead of silently not working.
- Only provide a clock when we are actually able to do so.
- Various small cleanups and compiler hints.
-
-2006-07-06 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Lutz Mueller <lutz at topfrose de>
-
- * gst/typefind/gsttypefindfunctions.c: (html_type_find),
- (plugin_init):
- Add typefinding for text/html (#346581).
+2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2006-07-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ * gst/subparse/qttextparse.c:
+ * gst/subparse/qttextparse.h:
+ subparse: add qttext support
+ Adds basic support for qttext subtitles, still lacks markup tags
+ to make it prettier, but the plain text already works.
+ Implemented according to:
+ http://www.apple.com/quicktime/tutorials/texttracks.html
+ http://www.apple.com/quicktime/tutorials/textdescriptors.html
+ Fixes #603357
+
+2009-12-01 13:22:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
- (xml_check_first_element), (xml_type_find), (smil_type_find):
- Fix SMIL typefinding, make xml_check_first_element() more
- useful.
+ * gst/subparse/gstsubparse.c:
+ subparse: conditionally cleanup sami context
+ Only cleanup sami context if we are parsing sami subtitles,
+ otherwise we might have crashes.
-2006-07-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (gst_play_base_bin_finalize), (decodebin_element_added_cb),
- (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
- * gst/playback/gstplaybasebin.h:
- Protect list of elements with a subtitle-encoding property and
- the subtitle encoding member itself with a lock of their own
- instead of using the object lock. This prevents a dead-lock in
- the element-remove callback in some circumstances when shutting
- down playbin.
+ * gst/subparse/gstsubparse.c:
+ subparse: Add missing caps to sink caps template
+ Some caps were missing from the sink caps template when
+ xml was disabled
-2006-07-05 Sebastien Moutte <sebastien@moutte.net>
+2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgsttag.def:
- Export some new functions.
- * win32/vs6/libgstogg.dsp:
- Add a link to libgsttag-0.10.lib.
+ * common:
+ Automatic update of common submodule
+ From 87bf428 to 47cb23a
-2006-07-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Some const-ification.
+ * common:
+ Automatic update of common submodule
+ From da4c75c to 87bf428
-2006-07-04 Wim Taymans <wim@fluendo.com>
+2009-11-30 10:22:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
- Improve checking if we are dealing with a stream. Added some
- more uris that need buffering.
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Fix some pad refcount issues
+ Fixes bug #603345.
-2006-07-03 Edward Hervey <edward@fluendo.com>
+2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
- Remove unused variable.
+ * common:
+ Automatic update of common submodule
+ From 53a2485 to da4c75c
-2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
- * Makefile.am:
- include lcov.mak
- * configure.ac:
- add GCOV_LIBS to GST_LIBS
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: handle theora streams with 0 keyoffset
-2006-07-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
- Patch by: Michael Sheldon <webmaster at mikeasoft com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle unknown streams
- * ext/alsa/gstalsasrc.c:
- Add 32 bps to template caps and increase channels range
- from [1,2] to [1,MAX]. See #346326.
+2009-11-26 14:30:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-30 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ Revert "textoverlay: First draw outline text and then the real text"
+ This reverts commit 60aa09d28c1f9fd29b56876d7ac6c0366d6cef4d.
+ First drawing the real text and then the outline produces ugly
+ text in lower resolutions. The outline line width needs to be somehow
+ changed relative to the resolution. Fixes bug #602924.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Recognise 'WMVA' video codec fourcc (#345879).
-
-2006-06-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- Fixed nasty memory leak
+2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
+ ...and fix code style a bit.
- * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
- (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
- fix logging
+2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-23 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ audiofilter: Add _CAST variants of the cast macros
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
- (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
- Protect remove_fakesink using a mutex, so that we don't try and
- remove the fakesink simultaneously from multiple threads.
+2009-11-25 10:26:16 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
- When going from READY to PAUSED, restore the fakesink, so that
- it is there when decodebin gets reused.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: add adjustement when slaving
+ Our calibration against the pipeline clock is done with the adjusted
+ ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
+ when reusing audio sinks after switching clocks and slaving methods in a
+ pipeline.
-2006-06-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertppayload.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/videorate/gstvideorate.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * sys/v4l/gstv4ljpegsrc.c:
- * sys/v4l/gstv4lmjpegsink.c:
- * sys/v4l/gstv4lsrc.c:
- * tests/examples/seek/scrubby.c:
- * tests/examples/seek/seek.c:
- Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
+ ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
+ Fixes bug #602834 and #350748.
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
- Second field in GEnumValue shouldn't be a description,
- but a stringified version of the enum value.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Reset last_granule during seeking
+ Fix case where we would reconstruct the wrong granulepos for
+ outgoing streams immediately after a seek.
-2006-06-22 Wim Taymans <wim@fluendo.com>
+2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
- (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
- Avoid type checking in buffer casts.
- Avoid caps copy in buffer_alloc when we can.
- Use pad_peer_accept.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Fix timestamp generation for theora
+ Timestamp generation was broken by the last commit for formats
+ with a non-zero granule shift. Also keep track of the last keyframe
+ so that we can regenerate granulepos for theora.
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/tag/tag.h:
- Oops, make that 'Since: 0.10.9'.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ * ext/ogg/vorbis_parse.c:
+ oggdemux: Fix vorbis parsing
+ Add a granule to granulepos conversion function. Fix the duration
+ function for vorbis. Handle timestamps on header packets differently
+ and be more careful about calculating OFFSET and OFFSET_END. After
+ this change, timestamps for vorbis don't exactly match up with the
+ timestamps that vorbisparse outputs, but it's unclear if vorbisparse
+ is actually correct and it would add a lot more code to make oggdemux
+ match vorbisparse. Fixes #602790.
+
+2009-11-19 19:28:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Transform QoS events to be meaningful for upstream elements
+ This is necessary because the sinks don't notice the group switches
+ and the decoders/demuxers have a different running time than the
+ sinks.
+ Fixes bug #537050.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
- (gst_tag_image_type_get_type):
- API: add GstTagImageType enum to describe images contained
- in image tags (#345641).
+2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ ogg: Fix generation of timestamps and durations
+ After changing some internal functions, I forgot to update
+ the code that puts the values on the buffers.
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Fix warnings with gst-inspect: "buffers-min" property
- should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
- typo in property description.
+2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/Makefile.am:
+ * ext/ogg/dirac_parse.c:
+ * ext/ogg/dirac_parse.h:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ * ext/ogg/vorbis_parse.c:
+ ogg: Add ogg stream parsing
+ Adds code that parses headers of various formats encapsulated in
+ Ogg in order to calculate timestamps and durations of each buffer.
+ Removes the creation of helper decoder elements to do this calculation
+ via conversion queries.
+ Fixes: #344013, #568014.
+
+2009-09-04 00:11:38 -0700 David Schleef <ds@schleef.org>
- Patch by: Cody Russell <bratsche at gnome org>
+ * ext/ogg/gstoggmux.c:
+ oggmux: don't overwrite object properties
- * gst/audioresample/gstaudioresample.c:
- (gst_audioresample_class_init):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_class_init):
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- Avoid unnecessary class cast check in class_init
- functions (#337747).
+2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-06-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoradec.c:
+ debug: also cast packet.packetno to gint64 in debug log
+ We do this already for granulepos to handle ogg_int64_t mismatches.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
- (gst_text_overlay_video_chain):
- g_markup_escape_text() REALLY doesn't like non-UTF8 input
- and doesn't validate its input either (and neither did
- textoverlay it seems). Let's do that then and fix #345206.
+2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-06-19 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
- (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
- (find_syncframe), (find_limits), (assign_value),
- (count_burst_unit), (gst_multi_fd_sink_new_client),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
- (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
- (gst_multi_fd_sink_change_state):
- * gst/tcp/gstmultifdsink.h:
- Added shiny new burst-on-connect methods.
- Add properties to control the minimal amount of data queued.
- Small cleanups.
- API: bytes-min property
- API: time-min property
- API: buffers-min property
- API: burst-unit property
- API: burst-value property
- API: add-full signal
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ debug: fix format string that was missing a var
- * gst/tcp/gsttcp-marshal.list:
- Added new marshaller code for the new signal.
+2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/elements/multifdsink.c: (GST_START_TEST),
- (multifdsink_suite):
- Added testcases for new burst methods.
+ * gst/adder/gstadder.c:
+ * tests/check/elements/adder.c:
+ adder: make events succeed, if they succed on atleast one pad
-2006-06-19 Edward Hervey <edward@fluendo.com>
+2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
- Implement clipping for accurate seeking.
- Closes #345225
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: error when all streams have no buffers
+ In some cases (all buffers dropped by a parser) a decodebin2
+ chain might receive an EOS before it gets enough data to
+ expose a decoded pad. In the case that no streams can expose
+ a pad we should error out instead of hang.
+ Fixes #542758
-2006-06-19 Wim Taymans <wim@fluendo.com>
+2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix stupid bug introduced in last commit
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
- (gst_video_scale_transform):
- Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
+2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-17 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Aggregate the stream-changed message by looking at the seqnum
+ Just counting how many messages were sent and how many were received
+ is not good enough because they might've been duplicated (e.g. by the
+ visualization audio tee). Comparing the sequence numbers should give
+ better results in that case.
- * configure.ac:
- Fix --disable-external (can't set conditionals conditionally,
- #343602).
+2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Ignore async state changes of the uridecodebins
+ Otherwise the async state change from READY->PAUSED of the
+ uridecodebins will take playbin2 from PLAYING->PAUSED again
+ during gapless group switches.
+ Fixes bug #602000.
- * tests/check/elements/audioresample.c: (test_reuse),
- (audioresample_suite):
- Add test case for bug #342789 fixed below.
+2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 0702fe1 to 53a2485
- * gst/audioresample/gstaudioresample.c:
- (gst_audioresample_class_init), (gst_audioresample_init),
- (audioresample_start), (audioresample_stop),
- (gst_audioresample_set_property), (gst_audioresample_get_property):
- Implement GstBaseTransform::start and ::stop so that audioresample
- can clear its internal state properly and be reused instead of
- causing non-negotiated errors with playbin under some circumstances
- (#342789).
+2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * tests/check/elements/audioresample.c: (setup_audioresample),
- (cleanup_audioresample):
- Need to set element state here so that ::start and ::stop are
- called.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: set to buffer less on no-more-pads
+ When a decodebin2 receives no-more-pads of a group it
+ can set that group's multiqueue buffering thresholds to
+ 'playing' buffering method, avoiding that it buffers
+ too long and cause problems when using with queue2.
+ See the associated bug for details.
+ Fixes #600787
-2006-06-16 Wim Taymans <wim@fluendo.com>
+2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: fix initial calibration
+ When we are calibrating the internal clock against the external clock take into
+ account the time offset applied to our internal clock because we will subtract
+ that in the render_function again.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
- Parse extra data better, apparently it's right behind
- the normal strf header size. Fixes #343500.
+2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-16 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't handle DURATION queries during group switches
+ During a group switch return the cached duration of the old group
+ because the old group still didn't finish playback. If we have no
+ cached duration return FALSE.
+ Fixes bug #585969.
- * ext/alsa/gstalsasink.c: (set_hwparams):
- If we fail to set the buffer_time and period_time alsa
- parameters, post a warning and leave alsa select a
- default instead of failing. Fixes #342085
+2009-11-15 19:36:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Post a stream-changed message after activating a group
+ This is useful to detect when playbin2 has really switched to the next
+ group after about-to-finish for example.
+ Fixes bug #584987.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/cdda/gstcddabasesrc.h:
- Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
- out in the header file and shouldn't be listed in the docs.
+2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
- Must dereference pointer to fourcc in the debug statement.
+ * win32/common/libgstvideo.def:
+ win32: Add new still-frame API to the defs
+ Add gst_video_event_new_still_frame() and
+ gst_video_event_parse_still_frame() functions to the win32 defs files
-2006-06-16 Stefan Kost <ensonic@users.sf.net>
+2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- add remaining symbols into correct setions
-
- * gst-libs/gst/audio/gstringbuffer.c:
- fix incomplete docs
-
- * gst-libs/gst/audio/gstringbuffer.h:
- comment out not yet implemented function
-
-
- * gst-libs/gst/floatcast/floatcast.h:
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- add short descriptions
-
-
- * gst-libs/gst/interfaces/propertyprobe.c:
- fix return value docs
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
- simplify debug logging
-
- * gst-libs/gst/riff/riff-read.h:
- sync function prototype and docs
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- remove left over symbol
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix 'uninitialized' compiler warning
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * autogen.sh:
* configure.ac:
- * docs/Makefile.am:
- Use GST_PLUGIN_DOCS macro in configure.ac, add
- --enable-plugin-docs default to autogen.sh and use
- ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
-
-2006-06-15 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
- (gst_ogg_demux_loop):
- Combine GstFlowReturn from the source pads to give a
- meaningfull result to the upstream peer or to stop the
- processing task in case of errors.
-
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gststreaminfo.c: (cb_probe):
- Try GST_TAG_CODEC as fallback when extracting the
- codec name; more debug info.
-
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
+ configure: bump core requirement to 0.10.25.1
+ We depend on new API that's only in git so far.
- * ext/ogg/Makefile.am:
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Extract language tags from ogm subtitle streams, so that
- the subtitle menu choices are labelled correctly in
- Totem (fixes #344708).
+2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-06-14 Wim Taymans <wim@fluendo.com>
-
- Patch by: Alessandro Decina <alessandro at nnva dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
- (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
- (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
- (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
- Fix various leaks. Fixes #343699.
- Add x-smoke mime type.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ * tests/check/libs/video.c:
+ video: Add functions to create/parse still frame events.
+ Add a new video event to mark the start or end of a still-frame
+ sequence, and a parser function to identify and extract info from
+ such events.
+ API: gst_video_event_new_still_frame()
+ API: gst_video_event_parse_still_frame()
+ Fixes: #601942
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-17 16:39:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- Add IDs for 'bext' chunks (see #343837).
+ * gst/playback/gstplaysink.c:
+ playsink: make sure we always go to PAUSED async
+ Set the need_async_start flag before going to PAUSED so that we always post the
+ ASYNC_START message, even after reusing playsink.
-2006-06-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-17 16:37:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/playback/gstplaysink.c:
+ playsink: make sure we remain a sink
+ When we remove our elements, we could lose our sink flag. Make sure we remain a
+ sink by setting the flag again after removing elements.
- * gst/subparse/samiparse.c: (sami_context_pop_state),
- (handle_start_font), (end_sami_element):
- Honour font face tags in SAMI subtitles (#344503).
+2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: remove unused array
- * po/POTFILES.in:
- add missing files containing translatable strings
+2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst/subparse/gstsubparse.c:
+ subparse: Use new double->fraction transformation function from core
- * docs/libs/tmpl/.cvsignore:
- we don't want those *.sgml files in CVS either
+2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Make subtitle error handling more robust and ignore late errors too
+ Make sure, to only "simulate" subtitle no-more-pads if it was still
+ pending and also handle errors in the subtitle pipeline as warnings
+ after the subtitles prerolled.
+ Don't set the suburidecodebin to READY after errors, handle_message
+ will usually be called from the streaming thread and doing that
+ from there is obviously not a good idea.
- * docs/libs/.cvsignore:
- * tests/check/elements/.cvsignore:
- * tests/check/libs/.cvsignore:
- ignore more
+2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
- * docs/libs/Makefile.am:
- also commiting the changed Makefile.am (added more libs to the
- doc-build)
+2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't leak the GError and debug string when parsing error messages
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- first batch of reordering things, add index & hierarchy
+2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
-2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: avoid crashing on SIGPIPE
+ Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
+ avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
+ Fixes #601772
- * configure.ac:
- use GST_PKG_CHECK_MODULES, cleans up output
+2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Improve subtitle passthrough in uridecodebin
+ Now the caps property isn't set anymore for the subtitle caps
+ but instead in the autoplug-continue signal it is detected
+ if the caps belong to a supported subtitle stream.
+ This makes automatic use of newly installed plugins.
- * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
- Add support for burn:// URIs (#343385); const-ify things a bit,
- use G_N_ELEMENTS instead of hard-coded array size.
+2009-11-11 17:08:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Only recreate factory caps if necessary and cache them
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-11-10 18:27:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
- Fix up broken entities before passing them to libxml *sigh*.
- (#343303).
-
-2006-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Only update the factory list when the registry has changed
+ Also don't free the list every time we go to NULL.
- * configure.ac:
- back to TRUNK
+2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.8 ===
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Use gst_pad_get_caps_reffed()
-2006-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- releasing 0.10.8, "Moar gij ziet mij nie"
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Use new "silent" property instead of unlinking
+ This makes sure that subtitleoverlay still gets segment updates and
+ everything to pass on downstream. Without this segment problems happen.
-2006-06-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- * win32/common/config.h:
- 0.10.7.2 prerelease
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Update segments after pushing the events downstream
+ This makes sure that we don't apply segments twice downstream. Also
+ always send our newsegment events downstream.
-2006-06-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * docs/libs/tmpl/gsttuner.sgml:
- * docs/libs/tmpl/gstxoverlay.sgml:
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/gstringbuffer.c:
- * gst-libs/gst/interfaces/colorbalance.c:
- * gst-libs/gst/interfaces/mixer.c:
- * gst-libs/gst/interfaces/tuner.c:
- * gst-libs/gst/interfaces/xoverlay.c:
- move last template doc snippets to source code and delete them
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Add silent property to disable subtitles
+ This tries to disable subtitles in the overlay or renderer
+ and if that's not possible it goes into passthrough mode.
-2006-06-06 Michael Smith <msmith@fluendo.com>
+2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
- (theora_parse_drain_queue):
- Mark DELTA_UNIT on non-keyframes.
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Set the video framerate on parsers if possible
+ Fixes bug #599649.
-2006-06-03 Jan Schmidt <thaytan@mad.scientist.com>
+2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
- (gst_ring_buffer_samples_done):
- * gst-libs/gst/audio/gstringbuffer.h:
- Document better the fact that latency_time and buffer_time are values
- stored in microseconds, and not the usual GStreamer nanoseconds.
- Change the variables (compatibly) that store them from GstClockTime
- to guint64 to make it more clear that they're not storing clock times.
- Also, remove the bogus property description that says the user can
- specify -1 to get the default value, since that's never been the case.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Make fps a GstFraction typed property and use it properly
- When computing the default segment size for the ring buffer, make it
- an integer number of samples.
+2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
- When the sub-class indicates a delay greater than the number of
- samples we've written return 0 from the audio sink get_time method.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Add property for the video framerate
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (set_channel_positions),
- (get_float_mc_caps), (get_int_mc_caps):
- * tests/check/elements/audioresample.c:
- * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
- * tests/check/elements/videorate.c:
- * tests/check/elements/videotestsrc.c: (GST_START_TEST):
- * tests/check/elements/volume.c:
- * tests/check/elements/vorbisdec.c:
- * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
- Don't busy-wait in tests; this was causing test timeouts very
- frequently when running under valgrind.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Handle external subtitles better
+ First of all, make sure that suburidecodebin never
+ errors out because of not-linked in case external subtitles
+ are used but then subtitles are disabled.
+ And then make sure that external subtitles always start from
+ the correct position and are not racing until EOS if they
+ get unselected and selected again.
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/README:
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
- (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_client_queue_caps),
- (gst_multi_fd_sink_client_queue_buffer),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_render):
- * gst/tcp/gstmultifdsink.h:
- make multifdsink properly deal with streamheader:
- - streamheader is taken from caps
- - buffers marked with IN_CAPS are not sent
- - streamheaders are sent, on connection, from the caps of the
- buffer where the client gets positioned to
- - further streamheader changes are done every time the client
- will receive a buffer with different caps
- * tests/check/elements/multifdsink.c: (GST_START_TEST),
- (gst_multifdsink_create_streamheader):
- add tests for this
+ * gst/playback/gstplaybin2.c:
+ playbin2: Flush the subtitles before switching to a new subtitle stream
+ This makes sure that all currently shown subtitles disappear
+ and new ones can be shown as soon as possible.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- Reinstate limit on channel count. Vorbis does not define the meaning
- of > 6 channels, so they're just independent channels. Gstreamer
- currently has no mechanism to represent N independent channels.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Set subtitle caps as raw caps for the uridecodebins
+ This will make sure that no subparse is ever plugged and subtitleoverlay,
+ that subpicture streams are handled the same was as subtitles and that
+ subtitle renderers are used if available.
+ Fixes bugs #595123, #570753, #591662, #591706.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- Don't arbitrarily restrict channel counts and rate in vorbis.
- In terms of effects likely on real-world files, this fixes 96kHz
- playback of vorbis.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2/playsink: Remove everything related to subpicture streams
+ These will soon be handled the same way as subtitle streams.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float):
- More correct float->int conversion.
+ * gst/playback/gstplaysink.c:
+ playsink: Add a queue before subtitleoverlay
+ This will improve playback, and the same thing is done
+ for subpicture streams too.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
- Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
- value. Fixes g-critical on trying to play back ogg containing
- unknown codec.
+ * gst/playback/gstplaysink.c:
+ playsink: Use subtitleoverlay for subtitles
-2006-06-02 Wim Taymans <wim@fluendo.com>
+2009-11-02 07:43:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
- (setup_source):
- * gst/playback/gstplaybasebin.h:
- Make the subtitle detection work from any thread so we don't
- deadlock. Fixes #343397.
-
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ subtitleoverlay: Add to the docs
- * gst/volume/Makefile.am:
- Seriously, it's not *that* hard to get compilation right. Even
- a drunk can do it ! Add LIBOIL CFLAGS and LIBS
+2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-01 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.c:
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Add new element for generic subtitle overlaying
+ This autopluggs the required elements for parsing and rendering
+ different subtitle formats on a video stream.
+ Fixes bug #600370.
- * gst/volume/gstvolume.c: (volume_choose_func),
- (volume_update_real_volume), (gst_volume_class_init),
- (gst_volume_init), (volume_process_float), (volume_process_int16),
- (volume_process_int16_clamp), (volume_set_caps),
- (volume_transform_ip), (plugin_init):
- * gst/volume/gstvolume.h:
- rewrite the passthrough check, split _int16 and _int16_clamp, fix
- another property desc., remove unused param from process function
-
- * tests/check/elements/volume.c: (volume_suite):
- reactivate the passthrough test
+2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-06-01 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoradec.c:
+ theoradec: Keep timestamp from incoming buffer if it is valid
+ Fixes bug #601627.
- * ext/alsa/gstalsamixerelement.h:
- * ext/alsa/gstalsamixeroptions.h:
- * ext/alsa/gstalsamixertrack.h:
- * ext/gnomevfs/gstgnomevfssink.h:
- * ext/gnomevfs/gstgnomevfssrc.h:
- * ext/theora/gsttheoradec.h:
- * ext/theora/gsttheoraenc.h:
- * ext/theora/gsttheoraparse.h:
- * ext/vorbis/vorbisparse.h:
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- * gst/audioconvert/gstaudioconvert.h:
- * gst/audioresample/gstaudioresample.h:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
- * gst/playback/gststreamselector.h:
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.h:
- * gst/videorate/gstvideorate.h:
- * gst/videoscale/gstvideoscale.h:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/volume/gstvolume.h:
- * sys/v4l/gstv4ljpegsrc.h:
- * sys/v4l/gstv4lmjpegsink.h:
- * sys/v4l/gstv4lmjpegsrc.h:
- * sys/v4l/gstv4lsrc.h:
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h:
- * tests/old/testsuite/alsa/sinesrc.h:
- Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
+2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-31 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playback: Update factories list on every access if the registry has changed
+ This makes application's simpler because the element doesn't need to
+ go to NULL first to make use of newly installed plugins.
+ Fixes bug #601480.
- * ext/libvisual/visual.c: (gst_visual_reset),
- (gst_visual_sink_setcaps), (gst_visual_sink_event),
- (gst_visual_src_event), (get_buffer), (gst_visual_chain):
- Handle DISCONT.
- Use running time before doing QoS.
- Handle mono too.
+2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-31 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playback: When going from NULL->READY check if the registry has new features
+ This makes it possible to use newly installed plugins after going back
+ to NULL instead of requiring a new instance.
+ Fixes bug #599266.
- * docs/libs/Makefile.am:
- set a magic variable to indicate we know the docs are incomplete
+2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-05-30 Sebastien Moutte <sebastien@moutte.net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clear the EOS state on a seek.
+ Allow seeking back into the stream after it hits EOS.
- * win32/common/libgstvideo.def:
- export gst_video_calculate_display_ratio
- * win32/vs6/libgstvideoscale.dsp:
- add link to libgstvideo-0.10.lib
+2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-30 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioresample/README:
+ * gst/audioresample/arch.h:
+ * gst/audioresample/fixed_arm4.h:
+ * gst/audioresample/fixed_arm5e.h:
+ * gst/audioresample/fixed_bfin.h:
+ * gst/audioresample/fixed_debug.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample_sse.h:
+ * gst/audioresample/speex_resampler.h:
+ audioresample: Update speex resampler to latest GIT
- * gst/playback/gstplaybasebin.c: (gen_source_element):
- Throw a more comprehensible error for rtsp:// URIs (rather
- than erroring out with a negotiation error later on) until
- we fix playbin to handle rtspsrc etc.
+2009-11-10 00:48:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-30 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playsink: assign chain->mute before using it
+ Fixes GObject warnings when starting totem.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
- (gst_text_overlay_text_event):
- Added some FIXMEs.
+2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
-2006-05-30 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoradec.c:
+ theora: Fix alignment of frames when converting
+ Fix logic inversion in calculating the offset in the theora
+ frame when copying to a GStreamer frame.
- * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
- (gst_adder_request_new_pad), (gst_adder_release_pad):
- * gst/adder/gstadder.h:
- Implement release_request_pad.
- Make padcounter atomic.
+2009-11-09 19:58:20 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
- Added check for release_pad in adder.
+ * gst/playback/gstfactorylists.c:
+ playback: Fix the order in strcmp that I broke in previous commit.
-2006-05-30 Wim Taymans <wim@fluendo.com>
+2009-11-09 19:16:21 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
- Fix build again.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Reduce number of calls to gst_type_find_peek.
+ Shaves off a couple percents off typefinding
-2006-05-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
- (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
- (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
- (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
- (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
- add more debugging
- clean up printf formats for granulepos and serialno
+ * gst/playback/gstfactorylists.c:
+ playback: Avoid expensive API calls in tight loop.
+ We know we're dealing with GstPluginFeature.
-2006-05-30 Michael Smith <msmith@fluendo.com>
+2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (raw_caps_factory),
- (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
- (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
- (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
- (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
- * ext/vorbis/vorbisenc.h:
- Multi-channel caps negotiation, so we can do proper multichannel
- vorbis encoding, negotiated through audioconvert.
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Add unit test for property settings
+ Also includes a regression test for bug #601104.
-2006-05-30 Wim Taymans <wim@fluendo.com>
+2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/adder.c: (test_event_message_received),
- (test_play_twice_message_received), (GST_START_TEST),
- (adder_suite):
- Added check to show that #339935 is fixed with ongoing
- adder and collectpads fixes.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Never return a negative track number in get_uri()
-2006-05-29 Wim Taymans <wim@fluendo.com>
+2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_request_new_pad):
- Don't leak pad name.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Don't set the track to 1 every time a device is set
+ Fixes bug #601104.
-2006-05-29 Wim Taymans <wim@fluendo.com>
+2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_query_duration),
- (forward_event_func), (forward_event), (gst_adder_src_event):
- Fix adder seeking.
- Make query/seeking code threadsafe.
+ * gst/playback/gstinputselector.c:
+ inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
- * tests/check/Makefile.am:
- * tests/check/elements/adder.c: (test_event_message_received),
- (GST_START_TEST), (test_play_twice_message_received):
- Fix adder test case.
+2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Add property to disable/enable posting of stream-topology messages
+ Most people don't need this messages and generating them is quite
+ expensive.
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (gst_play_base_bin_dispose),
- (set_encoding_element), (decodebin_element_added_cb),
- (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
- * gst/playback/gstplaybasebin.h:
- Add 'subtitle-encoding' property to playbin, so applications can
- force a subtitle encoding for non-UTF8 subtitles (#342268).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Protect subtitle elements and subtitle encoding by a new mutex
+ Using the object lock here can and will lead to deadlocks because
+ of deep-notifies of property changes: the deep-notify handler will
+ get the parent of objects, which will take the object lock again.
+ Fixes bug #600479.
- * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
- (gst_sub_parse_set_property):
- Rename recently-added 'encoding' property to 'subtitle-encoding'
- (so it can be proxied by playbin/decodebin in a generic way
- with less danger of false positives).
+2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-29 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstinputselector.c:
+ inputselector: Make sure that running_time->timestamp calculation never becomes negative
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
- (append_with_other_format), (set_structure_widths),
- (gst_audio_convert_transform_caps):
- Patch from #341562: give more specific audio caps in get_caps, so
- that basetransform can make better decisions on what caps to
- negotiate.
+2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ examples: Correct casting of g_signal* funcs first arguments
+ This completes the deprecated GTK API fix in commits 81a0a986 and
+ 79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
+ co take a gpointer, not a GtkObject.
- * tests/check/elements/volume.c:
- make it compile again
+2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Improve all-raw-caps detection for pads
- * tests/check/elements/volume.c: (volume_suite):
- disable test until #343196 gets resolved
+2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ basesrc: fix startup position in the ringbuffer
+ When we start and we need to produce the first sample, go to the next sample
+ that will be written into the ringbuffer instead of trying to go to sample 0.
+ We relied on rather small ringbuffer sizes to correctly go to the current
+ sample, which breaks whith large buffers.
+ Fixes #600945
- * gst/adder/gstadder.c: (gst_adder_get_type):
- Make it easier to copy&paste
-
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (volume_update_real_volume),
- (gst_volume_set_volume), (gst_volume_set_mute),
- (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
- (volume_transform_ip), (volume_update_mute),
- (volume_update_volume):
- * gst/volume/gstvolume.h:
- Add own debug category, move duplicate code to helper function, fix
- property texts, add more comments and prepare ffor liboil-goodness
-
- * tests/check/Makefile.am:
- * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
- add test for mute and passtrough case, be a bit more verbose to track
- failure
-
- * tests/check/generic/states.c: (GST_START_TEST):
- catch elements that fail to instantiate
+2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-28 Edward Hervey <edward@fluendo.com>
+ * gst/playback/gstinputselector.c:
+ inputselector: Use the start time (i.e. timestamp) as the last stop
+ Using the end time makes it impossible to replace buffers, which is
+ a big problem for subtitles that could have very long durations.
- * tests/check/pipelines/simple-launch-lines.c:
- * tests/check/pipelines/theoraenc.c:
- * tests/check/pipelines/vorbisenc.c:
- Comment out tests using parse_launch() if core was built without
- parsing capabilities.
+2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-27 Edward Hervey <edward@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Synchronize video/text based on the running time
+ Instead of simply using the buffer timestamps.
- * tests/check/Makefile.am:
- Extra bonus points for whoever explains to ensonic that you are meant
- to test unit tests thoroughly before commiting them, especially if
- you know it's going to break.
- De-activated element/adder tests.
+2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-27 Edward Hervey <edward@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Clip text buffers to the text segment and reset segments properly
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
- Marking caps conversion issues as GST_WARNING is way too verbose,
- Moving them to GST_LOG.
+2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-27 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Put the video segment into the instance struct instead of allocating it separately
- * README:
- Replace current README (containing the release notes from
- some 0.9.x version) with a proper README taken from the core.
+2009-11-06 09:05:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-26 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
- (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- Small cleanups.
- Add some FIXMEs
- Clip output samples to segment boundaries.
-
-2006-05-26 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
- Improve the errors produced on bad output, including some human
- readable description strings.
- Handle the (theoretical for ximagesink) case where the XServer
- has a different idea about the size required for a particular
- frame and gives us too small a memory allocation.
-
-2006-05-26 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
- Improve the errors produced on bad output, including some human
- readable description strings.
- Handle RGB Xv formats properly by transforming them into our
- big-endian caps description.
- Use gst_caps_truncate to ensure that we never try and choose a
- non-fixed caps in buffer_alloc.
- Handle the case where the XServer has a different idea about the size
- required for a particular frame and gives us too small a memory
- allocation.
- Use -1 to indicate 'no image format', because 0 is a valid XServer
- image format number.
- Put RGB Xv formats at the end of the caps, so that we always prefer
- YUV format frames.
- Iterate the available Xv Encodings to determine the maximum width and
- height, and then return that in our caps.
- (Closes #315312, #337544)
-
-2006-05-25 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
- When there is only one unfinished pad and it receives an event that
- doesn't match our requirements, we need to set alldone=FALSE so that
- the fakesink is not removed yet.
-
-2006-05-25 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- Use gst_type_find_helper_for_buffer() to find the type
- of stream from the first packet.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Check if text timestamp/duration is valid before clipping
- * configure.ac:
- Bump requirements to core CVS (needed for vorbis
- typefinding to work).
+2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-24 Edward Hervey <edward@fluendo.com>
+ * ext/theora/theoradec.c:
+ theoradec: printf format fix
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- Added the 'prfl' atom type which MQV (no, it's not a typo) files
- contain. Else they play perfectly fine with qtdemux.
+2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-05-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/gdp/gstgdpdepay.c:
+ gdpdepay: Clear adapter on flush and state change
+ Fixes #600469
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
- * gst/audiorate/gstaudiorate.c:
- make more debug catagories static
-
- * tests/check/Makefile.am:
- * tests/check/elements/adder.c: (message_received),
- (test_event_message_received), (GST_START_TEST),
- (test_play_twice_message_received), (adder_suite):
- added test case for using element twice, extra bonus points for anyone
- who can make these test run reliably
+2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstinputselector.c:
+ inputselector: use _get_caps_reffed()
- * ext/theora/theoradec.c: (theora_dec_chain):
- Make work with time-stamped input buffers that do not
- have a granulepos in BUFFER_OFFSET_END (like theora
- buffers coming from matroskademux). Fixes #342448.
+2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ pad: rename new api from _refed to _reffed.
+ Due to popular demand rename the new api as we still can.
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/Makefile.am:
- fdstresstest doesn't need Gtk+, fix compilation if
- gtk is not available (#342566).
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2: avoid copying caps
+ Use get_caps_refed() when we can.
-2006-05-19 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- 80 line columns
- Removed redundant floor()
+2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: use new getcaps function to avoid copies
+ Use the gst_pad_get_caps_refed() to avoid some caps copy functions.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
- On second thought, just skip JUNK chunks automatically, so
- the caller doesn't have to handle this. Fixes #342345.
- Also, return GST_FLOW_UNEXPECTED if we get a short read,
- not GST_FLOW_ERROR.
+2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use faster element_link_pads
+ Use the faster gst_element_link_pads because we know for sure the sinkpad name
+ and we don't need to have the function search for a suitable pad anymore.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
- Don't bail out on JUNK chunks with a size of 0 (would try to
- pull_range 0 bytes before, which sources don't like too much).
- See #342345.
+2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: make drift tolerance configurable
+ Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
+ drift or timestamp drift instead of relying on the latency-time value for clock
+ drift and 500ms for timestamp drift.
+ Remove warning about discont timestamp and simply resync. The warning is in some
+ cases not correct and is triggered more frequently now that we lower the
+ tolerance value.
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- Use the gstutil scaling function to preserve 64 bits while calculating
- output width and height from the display-aspect-ratio. (A continuation
- of #341542)
+2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Return NOT_LINKED for unselected text pads from a demuxer
+ We want to return NOT_LINKED for unselected pads but only for pads
+ from the normal uridecodebin. This makes sure that subtitle streams
+ are not raced past audio/video from decodebin2's multiqueue.
+ For pads from suburidecodebin OK should always be returned, otherwise
+ it will most likely stop with an error.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.h:
- When performing buffer allocations, remember the caps and image format
- we return so that if the same caps are asked for next time we can
- return them immediately without doing any caps intersections.
+2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
-2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
- * gst-libs/gst/rtp/README:
- Some new documentation
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
- New RTP audio base payloader class. Supports frame or sample based
- codecs. Not enabled in Makefile.am until approved.
+ * gst/playback/gstinputselector.c:
+ inputselector: also add inline to the proto to fix the build
+ Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/alsa.c: (test_device_property_probe):
- Fix test case: don't try to free NULL GValueArray when there
- are no devices.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Initialize caps property with the default raw caps
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/elements/alsa.c: (test_device_property_probe),
- (alsa_suite), (main):
- Add simple test that runs a device property probe on alsasrc,
- alsasink and alsamixer. Disable valgrind check for now (too
- many leaks in libasound, and valgrind ignored my suppressions
- additions).
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstrawcaps.h:
+ decodebin2: Use static caps for the default raw caps and put them into a separate header
+ This way we can use the same default raw caps everywhere.
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-03 08:26:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
- (gst_alsa_device_property_probe_probe_property),
- (gst_alsa_device_property_probe_needs_probe),
- (gst_alsa_device_property_probe_get_values),
- (gst_alsa_type_add_device_property_probe_interface):
- * ext/alsa/gstalsadeviceprobe.h:
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_init_interfaces):
- * ext/alsa/gstalsamixerelement.h:
- Clean up and simplify alsa device probing. Make it actually work
- for multiple classes. Don't cache results any longer.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: First draw outline text and then the real text
+ Improves the output a bit because no parts of the outline are
+ overwritten again.
- * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
- (gst_alsasink_init):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
- (gst_alsasrc_interface_supported), (gst_implements_interface_init),
- (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
- Make alsasink and alsasrc implement the GstPropertyProbe interface
- for device probing (#342181).
- Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
+2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin.c:
+ playbin: Make sure to keep a reference on the volume element
+ Fixes null pointer dereferences under certain circumstances.
+ Fixes bug #595401.
- * gst/subparse/samiparse.c: (handle_start_font):
- Don't ignore return value of strtol (++compiler_happiness).
+2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-05-17 Tim-Philipp Müller <tim at centricular dot net>
+ * po/POTFILES.in:
+ po: queue2 has moved to core
- Patch by: Young-Ho Cha <ganadist chollian net>
+2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_class_init), (gst_sub_parse_init),
- (gst_sub_parse_set_property), (gst_sub_parse_get_property),
- (convert_encoding):
- * gst/subparse/gstsubparse.h:
- Add 'encoding' property (#341681).
+ * gst/playback/gstplaysink.c:
+ playsink: Reset {mute,volume}-changed flags after setting the volume
+ These flags are there to make sure that the volume is set, if there
+ is no volume element yet.
- * gst/subparse/samiparse.c: (characters_sami):
- Output is pango markup, so we need to escape text
- between tags (#342143).
+2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions):
- It's okay to have caps with channels=1 and a channel position
- different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
- (deinterleavers might want to keep the position in the caps,
- so that they can be re-interleaved again properly later).
- Leave check for unexpected 2-channel layouts intact for now.
+2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/playback/gstplaysink.c:
+ playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
+ Fixes bug #600027.
- * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
- Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
- basesrc can do its job correctly.
+2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
- (gst_alsa_detect_formats), (get_channel_free_structure),
- (caps_add_channel_configuration), (gst_alsa_detect_channels),
- (gst_alsa_probe_supported_formats):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- Refactor and improve caps probing code: probe signedness
- when we probe the supported formats/widths; set endianness
- to the one we actually probed for (ie. cpu endianness).
-
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
- (gst_alsasrc_close):
- * ext/alsa/gstalsasrc.h:
- Implement caps probing for alsasrc.
+2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-15 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (gst_theora_dec_reset),
- (theora_dec_src_query), (theora_dec_src_event),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_data_packet), (theora_dec_change_state):
- Cleanups, add some G_LIKELY.
- Use segment helpers instead of our own wrong code.
- Clear queued buffers on seek and READY.
-
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_sink_event),
- (vorbis_handle_comment_packet), (vorbis_dec_push),
- (vorbis_handle_data_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- * ext/vorbis/vorbisdec.h:
- Remove old useless packetno variable.
- Do position query properly.
- Add some G_LIKELY.
- Do cleanup of queued buffers in new helper function
- and use it.
+ * docs/plugins/inspect/plugin-queue2.xml:
+ queue2: Remove inspect file
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- Query supported sample rates. Fixes #341732.
+ * gst/playback/Makefile.am:
+ * gst/playback/gstqueue2.c:
+ queue2: Remove from gst-plugins-base
+ This is now in coreplugins.
-2006-05-15 Julien MOUTTE <julien@moutte.net>
+2009-10-28 11:29:36 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c: (cleanup_decodebin),
- (gst_decode_bin_change_state): Make decodebin reusable
- when going from PAUSE_TO_READY and then back to PAUSED.
- Fixes #331678.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ docs: include more indexes
-2006-05-15 Wim Taymans <wim@fluendo.com>
+2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_sink_query), (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
- (vorbis_dec_clean_queued), (vorbis_dec_push),
- (vorbis_handle_data_packet), (vorbis_dec_change_state):
- Cleanups. Use refcounting and DEBUG_OBJECT.
- Reset segment on flush, use code methods instead of our
- own wrong version.
- Fix potential memleak.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ docs: turn entities into xi:includes
+ This is faster to process and easier to maintain. Its also less 80s.
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
- (gst_alsasink_init):
- * ext/alsa/gstalsasink.h:
- Don't leak allocated snd_output_t structure if there's
- more than one alsasink instance at a time (#341873).
- Also fix GObject macros in header file.
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: dump packets which we reject
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't use libxml functions in the typefinding code.
+2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-15 Wim Taymans <wim@fluendo.com>
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore basetime unit test binary
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
- Fix seeking performance in the case where a non-header
- packet has a 0 granulepos (busted theora case).
- Fixes #341719
- (nvdr: thanks Wim for not mentioning who spotted and fixed it in
- the first place, you limelight stealer you)
+2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst/adder/gstadder.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstqueue2.c:
+ * gst/playback/gststreaminfo.c:
+ * gst/playback/gststreamselector.c:
+ * gst/subparse/gstssaparse.c:
+ Remove GST_DEBUG_FUNCPTR where they're pointless
+ There's not much point in using GST_DEBUG_FUNCPTR with GObject
+ virtual functions such as get_property, set_propery, finalize and
+ dispose, since they'll never be used by anyone anyway. Saves a
+ few bytes and possibly a sixteenth of a polar bear.
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Improve SAMI typefinding: handle case where there are
- whitespaces or newlines in front of the first <SAMI>
- tag (#169936).
+2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstqueue2.c:
+ queue2: add custom acceptcaps function
- * configure.ac:
- Build video4linux plugin even if there's no XVIDEO, just
- without implementing the GstXOverlay interface (#334002).
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
- (plugin_init):
- Add tentative support for libvisual-0.4 (#336881).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement low/high watermark property
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * tests/examples/seek/seek.c:
+ seek: add checkbox to enable buffering
- * gst/subparse/samiparse.c: (handle_start_font):
- Need to map "silver" colour explicitly (#169936).
+2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: don't use 2 buffering elements
+ Only use the multiqueue buffering when we don't have a stream (and thus are
+ using queue2 to do the buffering already).
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (feed_textbuf),
- (gst_subparse_type_find), (plugin_init):
- * gst/subparse/gstsubparse.h:
- * gst/subparse/samiparse.c:
- * gst/subparse/samiparse.h:
- Add support for SAMI subtitles (#169936).
-
-2006-05-14 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ playbin2: add flag to enable decodebin buffering
+ Add a flag that enables buffering in decodebin.
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
- Fix #341696: crash when mixing L+R+C to mono or stereo.
- * tests/check/Makefile.am:
- * tests/check/elements/audioconvert.c: (set_channel_positions),
- (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
- (audioconvert_suite):
- Add test for the above, including some generic framework bits for
- testing multichannel things.
+2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.10.7 ===
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: buffering is implemented now
-2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.10.7, "Leave the gun"
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: buffering is implemented now
-2006-05-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- Fix the build.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: configure use-buffering on multiqueue
-2006-05-12 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
- * gst-libs/gst/video/video.h:
- * gst/videoscale/Makefile.am:
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- * tests/check/Makefile.am:
- * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
- (main):
- Fix integer overflow problem with pixel-aspect-ratio calculations
- in videoscale and xvimagesink (#341542)
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use 0 for max buffer size
-2006-05-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c:
- Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: set some reasonable defaults
-2006-05-12 Sebastien Moutte <sebastien@moutte.net>
+2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * win32/MANIFEST:
- update win32 files listing
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: set buffering properties on decodebin2
+ Propagate the buffering properties on decodebin2 but only if we are not already
+ doing download buffering.
-2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- patch by: Sjoerd Simons (sjoerd@luon.net)
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add use-buffering property
+ Add a use-buffering property that will perform buffering on the parsed or
+ demuxed media.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (group_create), (group_destroy), (add_stream),
- (gst_play_base_bin_get_property),
- (gst_play_base_bin_get_streaminfo_value_array):
- * gst/playback/gstplaybasebin.h:
- API: GstPlayBaseBin::stream-info-value-array property
- use a more bindings-friendly way of exposing streaminfo
- using a GValueArray. Tested in ipython.
- Closes #341114
+2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-11 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: refactor queue size configuration.
+ Refactor the queue size configuration into a new method.
+ Use the same queue values for buffering as for preroll.
- * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
- (queue_underrun_cb), (queue_filled_cb):
- Also catch queue underruns but don't do anything yet.
- Refactor and comment queue enlarging code a bit.
+2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (queue_overrun),
- (queue_threshold_reached), (queue_out_of_data),
- (gen_preroll_element):
- If a queue over/underruns check that we don't create nasty
- deadlocks when the min-threshold is not reached but the
- max-bytes is. In those cases disable max-bytes when we
- know that the queue is fed timed data.
- Add more comments.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: move error path down
-2006-05-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-23 11:02:40 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gen_audio_element):
- Make playbin automatically plug an 'audioresample'
- element before the audio sink as well. This solves
- problems with sinks that only accept a very specific
- sample rate, like esdsink (e.g. #340379).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement max queue size properties
-2006-05-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-23 10:42:23 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gen_source_element):
- Make http sources send special headers so that we receive
- icecast metadata if the http stream is an icecast stream
- (otherwise the server will just ignore them). This also
- means that from now on users will need the 'icydemux'
- element from gst-plugins-good installed if they want to
- listen to icecast radio streams. (#341432, #333657).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add properties for buffering
+ Add properties that can be used to configure the multiqueue buffers and
+ buffering methods
-2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
- remove stupid example from docs - it should come with a simple
- C program instead.
- Clean up/fix docs
- * tests/check/elements/multifdsink.c: (wait_bytes_served),
- (fail_if_can_read), (GST_START_TEST),
- (gst_multifdsink_create_streamheader), (multifdsink_suite):
- add a test for changing streamheader which exposes a bug in
- multifdsink
+ * tests/examples/app/Makefile.am:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/v4l/Makefile.am:
+ examples: fix linking order.
+ the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
+ path before the local ones... resulting in the example applications picking
+ up the system-wide libraries and not the (potentially modified) uninstalled
+ libraries
-2006-05-11 Michael Smith <msmith@fluendo.com>
+2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
- (gst_gnome_vfs_src_received_headers_callback):
- * ext/gnomevfs/gstgnomevfssrc.h:
- Don't set icy-caps unless we have a sane interval value. Move
- interval to a local variable; we never use it outside this function.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't destroy the suburidecodebin on errors
+ It can still be reused
-2006-05-11 Wim Taymans <wim@fluendo.com>
+2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
- Register special buffer types along with the objects so
- that they are not registered at runtime from N different
- streaming threads since they are not threadsafe.
+ * gst/playback/gstplaybin2.c:
+ playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/multifdsink.c: (wait_bytes_served),
- (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
- add two more tests, one doing streamheader
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't set uridecodebin states to NULL before reusing them
+ This makes sure that the internal decodebin2 and everything else can
+ be reused without reinstantiation.
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
- clean up the bufqueue when shutting down
- * tests/check/Makefile.am:
- * tests/check/elements/multifdsink.c: (setup_multifdsink),
- (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
- (main):
- add a test for the leak that was just fixed
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Store unused decodebin2 instances for further usage.
+ This allows faster re-use of uridecodebin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=599471
-2006-05-10 Wim Taymans <wim@fluendo.com>
+2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_query_duration), (gst_adder_query), (forward_event),
- (gst_adder_src_event), (gst_adder_sink_event),
- (gst_adder_class_init), (gst_adder_finalize),
- (gst_adder_request_new_pad), (gst_adder_collected):
- * gst/adder/gstadder.h:
- Updated some docs. Added comments and FIXMEs all over the place.
- Improve debugging info.
- Fix leak on finalize by not calling the parent.
- Implement duration query.
- Make event forwarding threadsafe.
- Correctly send NEWSEGMENT at start and after flush.
- Handle EOS correctly.
- Post error when not negotiated.
-
- * tests/check/elements/adder.c: (GST_START_TEST):
- Added FIXME in the test.
-
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
- (gst_text_overlay_halign_get_type),
- (gst_text_overlay_wrap_mode_get_type):
- * ext/theora/theoradec.c: (theora_handle_type_packet),
- (theora_handle_data_packet):
- * ext/theora/theoraenc.c: (gst_border_mode_get_type),
- (theora_enc_sink_setcaps), (theora_enc_chain):
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_mode_get_type):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
- * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_sync_method_get_type), (gst_unit_type_get_type),
- (gst_client_status_get_type):
- * gst/videoscale/gstvideoscale.c:
- (gst_video_scale_method_get_type):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_pattern_get_type):
- * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
- (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
- (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
- (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
- (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
- (paint_setup_RGB565), (paint_setup_xRGB1555):
- Const-ify GEnumValue and GFlagsValue arrays. Use
- GST_ROUND_UP_* macros instead of home-made ones.
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c:
+ theora: Convert theoraparse to libtheora 1.0 API
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
- * configure.ac:
- Require core CVS for the new newsegment stuff.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ rtpaudiopayload: Only sent exact multiple of the frame size
+ Also align the maximum size with the frame size, not only the minimum
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
- Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: move debug calculation into debug macro
+ Remove in_duration and move its calculation to
+ GST_LOG_OBJECT macro. This way it will only be calculated
+ if we have debug enabled.
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
- Register nick for enum value (#341160).
+2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
-2006-05-09 Stefan Kost <ensonic@users.sf.net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Removing unused variable
+ The in_stop variable was never read. Removing it.
- * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
- (plugin_init):
- backout typefind patch #340375
-
- * tests/check/elements/adder.c: (message_received),
- (GST_START_TEST), (adder_suite):
- redo, signal-handling of test
+2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
-2006-05-09 Wim Taymans <wim@fluendo.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: be more accurate on offset math
+ Replace gst_util_uint64_scale_int for its rounding version
+ to improve accuracy and avoid inserting samples where
+ they aren't needed.
+ Fixes #499181
- * gst/adder/gstadder.c: (gst_adder_request_new_pad),
- (gst_adder_collected):
- * gst/adder/gstadder.h:
- Remove bogus segment merging and forwarding, we don't
- care about timestamps anyway and we just produce a
- continuous stream.
- Also create a nice NEWSEGMENT event when we start.
- Use _scale_int some more.
+2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Edward Hervey <edward@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Optimize a bit more
+ ...and add a FIXME for bug #598695 and explain
+ what we should do once Pango supports user fonts.
- * tests/icles/stress-xoverlay.c:
- Fix if core was built without parsing support.
+2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c:
+ subparse: Add support for DKS subtitle format
+ Fixes bug #598936.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add SEDG (Samsung MPEG-4) fourcc.
+2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Edward Hervey <edward@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Do shading as first operation
- * tests/icles/stress-xoverlay.c:
- * tests/examples/volume/volume.c:
- Fix if core was built without parsing support.
+2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c:
- Disable the parse_launch example if core was built without parsing
- support.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only use a single cairo surface for drawing
+ ... and comment/optimize what is going on here a bit better.
-2006-05-08 Edward Hervey <edward@fluendo.com>
+2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * autogen.sh: (CONFIGURE_DEF_OPT):
- libtoolize on Darwin/MacOSX is called glibtoolize
+ * gst/playback/gstinputselector.c:
+ inputselector: set output caps before pushing
+ Set the output caps on the srcpad before pushing the buffer because else core
+ will do a rather expensive check to see if we can actually accept those caps on
+ the srcpad.
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
- Disable the adder test, until the build-slaves posses the kindness to
- either like it or to give valid reason for not doing so
+ * gst/playback/gstinputselector.c:
+ inputselector: install an acceptcaps function
+ Install a custom acceptcaps function instead of using the default expensive
+ check. We accept whatever downstream accepts so we pass along the acceptcaps
+ call to the downstream peer.
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
- (adder_suite):
- Shuffle NULL state change around and raise timeout more
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix typo in previous mxf typefinder change
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
- (mp4_type_find), (plugin_init):
- Add typefind to distinguish between "audio/x-m4a" and new type
- "video/mp4". Fixes #340375
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: speed up mxf_type_find over 300 times for worst case scenarios
+ * memcmp is expensive and was being abused, reduce calling it by checking
+ the first byte.
+ * iterating one byte at at time over 64 kbites introduces a certain overhead,
+ therefore we now do it in chunks of 1024 bytes
+ And I do mean over 300 times. The average instruction call per mxf_type_find
+ was previously 785685 and it's now down to 2458 :)
- * tests/check/elements/adder.c: (adder_suite):
- Raise timeout to make buildbot happy
+2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstfactorylists.c:
+ decodebin2: avoid type checks
- * gst/adder/gstadder.c: (gst_adder_sink_event),
- (gst_adder_request_new_pad), (gst_adder_change_state):
- * gst/adder/gstadder.h:
- * tests/check/Makefile.am:
- * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
- (adder_suite), (main):
- Add sink-event handling to adder. It tries to merge incomming
- newsegment-events. Added test to check if segment_done is comming
- through. Fixes #340060
+2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-05-05 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstdecodebin2.c:
+ gst/decodebin2: Ensure we get fixed caps for topology message
+ There are some corner cases (like with dvdemux amongst others) where
+ the caps won't be negotiated, but the pad has fixed caps.
- * ext/theora/theoraparse.c (gst_theora_parse_init)
- (theora_parse_src_convert, theora_parse_src_query):
- * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
- (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
- query functions on the source pads of the theora and vorbis parse
- elements. Fixes position querying when doing a remux.
+2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-05-05 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ gst/decodebin2: Don't expose chains if we're shutting down.
+ This avoids adding flushing pads to ourself
- * ext/theora/theoraparse.c: (parse_granulepos),
- (theora_parse_drain_queue_prematurely),
- (theora_parse_queue_buffer), (theora_parse_sink_event):
- Fix flushing.
- Fix invalid granulepos outputs when starting with a non-keyframe.
+2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ * ext/pango/gsttextoverlay.c:
+ pango: bump pango requirement to stable version and remove ifdefs
+ Bump pango requirement from an ancient development version to an
+ ancient stable version.
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
- (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
- Rearrange MPEG system stream detection, fixing some memleaks in the
- process.
- Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
- they clean up their data correctly.
- Remove unused ogganx caps and move the 'is_annodex' check to inside
- the 'is_ogg' if statement.
+2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-05 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/.gitignore:
+ .gitignore: update after files got renamed
- * gst/playback/gstdecodebin.c: (cleanup_decodebin):
- Properly remove ghostpads. Fixes #340392
+2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-04 David Schleef <ds@schleef.org>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: small comment fix
- * gst/typefind/gsttypefindfunctions.c:
+2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-05-03 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtp: Correct timestamping of buffers when buffer_lists are used
+ The timestamping of buffers when buffer_lists are used failed if
+ a buffer did not have both a timestamp and an offset.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg_ts_probe_headers), (mpeg_ts_type_find):
+2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
- When typefinding an MP3 in push-based mode, don't penalise the
- probability down to 74% when we found 5 valid frames just because we
- can't peek the end of the file.
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtsp-marshal.list:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ * gst-libs/gst/rtsp/rtsp-marshal.list:
+ * gst-libs/gst/video/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/tcp/Makefile.am:
+ build: fix previous commit to fully accomodate the glib-gen.mak changes
+ I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
+ marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
- Make the probability for detecting MPEG Transport Streams based on the
- number of sequential headers we successfully detected.
+2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-05-03 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/tcp/Makefile.am:
+ build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
+ The build rules in glib-gen.mak were using pattern rules in a non save way.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_dec_push), (vorbis_dec_chain):
- Still produce an error when we receive an empty packet.
+2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-05-03 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 85d1530 to 0702fe1
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
- (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
- Mark buffers with DISCONT after seek and after activating new
- chains.
+2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_reset),
- (theora_get_query_types), (theora_dec_sink_event),
- (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
- (theora_dec_change_state):
- Fix frame counter.
- Detect and mark DISCONT buffers.
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- * ext/vorbis/vorbisdec.h:
- Use GstSegment.
- Detect and mark DISCONT buffers.
- Don't crash on 0 sized buffers.
+ * ext/theora/theoradec.c:
+ theora: Make theoradec use gstvideo for image conversion
+ Vastly simplifies code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-05-03 Wim Taymans <wim@fluendo.com>
+2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
- * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
- (volume_transform_ip):
- Increase "volume" property to 10.0. Fixes #340369.
- Set the process function to NULL when capsnego fails so that
- we properly error out.
+ * ext/theora/theoradec.c:
+ theora: Don't always round to even width/height
+ Previously, the code always rounded to even sizes. Now it only ensures
+ that pic_x and pic_y are multiples of 2 if the output format requires
+ it.
+ Also inlcudes fixes to take pic_x/y into account properly when copying
+ the buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-05-02 Stefan Kost <ensonic@users.sf.net>
+2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
- * gst/playback/gstplaybin.c: (add_sink):
- * gst/playback/test.c: (main):
- * gst/playback/test5.c: (dump_element_stats):
- * gst/playback/test6.c: (main):
- free caps using gst_caps_unref, don't leak caps-strings
+ * configure.ac:
+ theora: Don't check for theora.pc anymore
+ THe new APIs from theoradec and theoraenc are used now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-05-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
- * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
- (plugin_init):
- Refine musepack typefinding a bit. Return MAXIMUM
- probability when we detect stream version 7 to make
- sure the mpeg audio typefinder doesn't trump us.
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: Convert theoradec to libtheora 1.0 API
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Protect against unexpected NULL strf_data buffer.
+ * ext/theora/Makefile.am:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Port encoder to new Theora API
+ Includes ripping out the old buffer copy code to fill up to frame size.
+ This is not necesary with the new encoder.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- interpret the out[] buffer in the order the bytes are actually
- put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
- Other tests should use BYTE_ORDER since the array is filled in
- with actual values
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Disable sharpness property
+ It's ignored by libtheora
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-09 21:57:08 +0200 Benjamin Otte <otte@gnome.org>
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- when a test fails, give an indication of which it is
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Disable noise-sensitivity property
+ It is ignored by libtheora
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-09 21:50:57 +0200 Benjamin Otte <otte@gnome.org>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- compile fix; strtol() needs <stdlib.h>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Disable keyframe-mindistance property
+ It's ignored by the current Theora library
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+2009-09-09 21:48:08 +0200 Benjamin Otte <otte@gnome.org>
- * ext/alsa/gstalsamixerelement.c:
- * ext/alsa/gstalsasrc.c:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/gnomevfs/gstgnomevfssink.c:
- * ext/gnomevfs/gstgnomevfssrc.c:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c:
- * ext/ogg/gstogmparse.c:
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/pango/gsttimeoverlay.c:
- * ext/theora/theoradec.c:
+ * ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/audio/gstaudiofilter.c:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audiorate/gstaudiorate.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreamselector.c:
- * gst/subparse/gstsubparse.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- * gst/videorate/gstvideorate.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- * sys/v4l/gstv4ljpegsrc.c:
- * sys/v4l/gstv4lmjpegsink.c:
- * sys/v4l/gstv4lmjpegsrc.c:
- * sys/v4l/gstv4lsrc.c:
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- * tests/check/libs/cddabasesrc.c:
- make GstElementDetails const
+ theora: Disable keyframe_threshold property
+ It's ignored by the current theora encoder
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
- * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
- (gst_adder_init):
- send events from src-pad to all sink-pads fixes #338657
-
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Get rid of "quick" property
+ The proeprty is not used by libtheora at all
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
- (alsasink_parse_spec):
- query width capabilities from alsa, fixes #338919
+2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * ext/theora/theoraenc.c:
+ theora: remove support for outdated granulepos hack
+ This is in preparation to switching to switching to the new Theora API
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_remove_client_link):
- * gst/tcp/gstmultifdsink.h:
- Fix race condition in multifdsink that can lead to spurious
- duplicate clients. this patch adds a new signal that is fired when
- multifdsink has removed all references to the fd.
- Fixes #339574.
- Updated documentation.
- API: client-fd-removed signal added
+2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
-2006-04-28 Michael Smith <msmith@fluendo.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Ignore border property
+ Always make the video use black as padding color.
+ The output will be identical to previous versions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
- When asking g_value_array_new to prealloc elements, we may as well
- ask for the right number of elements.
+2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Ignore the center property, always set video to top left
+ This is not a necessary property, the output will be identical no matter
+ what.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- patch to make timestamp checking more tollerant to rounding
- errors given that real discontinuities are to be marked on
- buffers. Fixes some asf files and #338778.
- Also avoid some crashers when we receive an event in the
- NULL state.
-
-2006-04-28 Michael Smith <msmith@fluendo.com>
-
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
- (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
- (gst_gnome_vfs_src_get_property),
- (gst_gnome_vfs_src_send_additional_headers_callback),
- (gst_gnome_vfs_src_received_headers_callback),
- (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
- (gst_gnome_vfs_src_stop):
- * ext/gnomevfs/gstgnomevfssrc.h:
- Remove ICY handling (mostly) from gnomevfssrc, in favour of
- proper shared support within icydemux.
+2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ * po/Makevars:
+ po: Don't create backup .po files
+ As well as preventing creation of useless backup files, it works
+ around a bug in gettext 0.17 on OS/X
- * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
- (gst_video_rate_swap_prev), (gst_video_rate_chain):
- fix up docs
- fix a leak when no caps negotiated
- fix counting of input frames
- * tests/check/elements/.cvsignore:
- * tests/check/elements/videorate.c: (assert_videorate_stats),
- (GST_START_TEST), (videorate_suite):
- add tests for these
+2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post a element message on the bus with the stream topology
+ Fixes bug #598533.
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
- (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (gst_ring_buffer_stop),
- (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
- (gst_ring_buffer_commit), (gst_ring_buffer_read),
- (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
- (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
- Check arguments passed to public functions instead of
- crashing.
+2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Store the "endcaps" of a chain
+ This are the caps that either resulted in a deadend if
+ no plugin for them could be found or raw caps.
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
- (gst_base_audio_src_get_time), (gst_base_audio_src_create):
- GstBaseAudioSrc must be live or it does not work.
+2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
- Don't set live to TRUE as this is the default in the parentclass.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Store for every chain, which pad resulted in its creation
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
- (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
- Videoscale doesn't pass on pixel-aspect ratio. Handle all
- fixation cases better. Fixes #338991
+ * tests/check/pipelines/basetime.c:
+ check: Don't fail the basetime test when no audiosrc is available
+ On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
+ it isn't in gst-plugins-base. Just defer the test, instead of
+ failing it.
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- Handle 0/1 framerate correctly Fixes #331901.
+ * common:
+ Automatic update of common submodule
+ From a3e3ce4 to 85d1530
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (get_float_caps),
- (GST_START_TEST), (audioconvert_suite):
- Added check for correct clipping when doing float samples
- in audioconvert.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
- * gst/videorate/gstvideorate.c: (gst_video_rate_event),
- (gst_video_rate_chain):
- Print more debugging info.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix timestamp comparission, Fixes #597407
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (resample_set_state_from_caps):
- Add support for other formats audioresample can handle such as
- 32 bits in and float and 64 bits float. Fixes #301759
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/basetime.c:
+ tests: new test for baseaudiosrc base_time comparison
+ This test reveals a bug in comparison operation between timestamp and
+ GstElement's base_time in GstBaseAudioSrc.
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-10-08 19:55:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float):
- correctly clip float samples > 1.0. Fixes #338718
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't stop completely on initialization errors from subtitle elements
+ Instead disable the subtitles and play the other parts of the stream.
+ Fixes bug #587704.
-2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Ignore no-more-pads from non-demuxer elements
+ instead of printing an error that no corresponding group could
+ be found. no-more-pads from non-demuxer elements doesn't give
+ any additional information because there can only be a single srcpad.
+ Fixes bug #598288.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
- (gst_text_overlay_render_text):
- Don't strip newlines from the text. Also, center lines
- within multi-line paragraphs (#339405).
+2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: track active conversion in perf log
- * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
- Fix wavpack typefinding to work in more cases (don't peek
- for chunks of multiple hundred kBs at once, but process
- things step-by-step in smaller units). Fixes #339786.
+2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: handle socket errors
+ gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
+ on a socekt. Fix this problem by checking for error on 'other' socket after poll
+ return.
+ Fixes #596159
- * configure.ac:
- back to HEAD
+2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.10.6 ===
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: whitespace fixes
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.10.6, "Chao"
+ * ext/theora/theoradec.c:
+ theoradec: avoid confusing error
-2006-04-26 David Schleef <ds@schleef.org>
+2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst/videoscale/gstvideoscale.c: Add call to oil_init().
- Fixes #338897.
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: fixes warings in macosx snow leopard
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
- * configure.ac:
- * win32/common/config.h:
- new prerelease
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraparse.c:
+ theora: fixes warnings on macosx snow leopard
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.com>
- patch by: Wim Taymans
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek):
- make sure correct newsegments are sent, so that the decoder
- and the demuxer agree on timestamps. Fixes playback of a lot
- of Ogg files that do not start from 0. Fixes #339833.
-
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Edward Hervey <edward@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
- * tests/check/Makefile.am:
- * tests/check/elements/videorate.c: (assert_videorate_stats),
- (setup_videorate), (cleanup_videorate), (GST_START_TEST),
- (videorate_suite), (main):
- Fix an infinite loop if frames are passed in with wrongly ordered
- timestamps. Fixes #339013.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- fix typefinding on some ISO files. Fixes #339212.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- add another H264 fourcc. Fixes #339047.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Jan Schmidt
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_bufferalloc):
- Restore old StreamSelector behaviour.
- Fixes #338419.
-
-2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * win32/common/config.h:
- prerelease
-
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
- Fix some memory leaks: on finalize, free buffers left in the queue
- before destroying the queue; in _push(), unref rtp_buf even if
- the process vfunc returned a NULL buffer as output buffer (#337548);
- demote some recuring debug messages to LOG level.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ ogg: fixes warnings on macosx snow leopard
-2006-04-11 Wim Taymans <wim@fluendo.com>
+2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
- (gst_ogg_demux_loop):
- More cleanups.
- Respect segment stop when emiting EOS or SEGMENT_DONE.
- Fixes (#337945).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix a warning in macosx
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_get_property):
- Don't leak pad name.
+ * gst-libs/gst/tag/tags.c:
+ tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
-2006-04-10 Michael Smith <msmith@fluendo.com>
+2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/check/Makefile.am:
* tests/check/gst-plugins-base.supp:
- Suppress an old libtheora bug (fixed in more recent versions), so
- that FC4 buildslaves can pass.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
- (gst_ogg_demux_init), (gst_ogg_demux_finalize),
- (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
- (gst_ogg_demux_loop):
- Don't leak events.
- Remember what error we got when finding chains, if we
- were shutdown, that would not be an error.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event):
- Starting the ringbuffer when we did not acquire it can cause
- a deadlock, is pointless and causes nasty things for
- subclasses.
- Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
+2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
- Add some more debugging.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ audioconvert: change the format instead of cast as ensonic asked
- * ext/theora/theoradec.c: (theora_dec_src_event),
- (theora_handle_data_packet):
- Some more debug info.
+2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
- * tests/examples/seek/seek.c: (start_seek), (main):
- Print element messages too.
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
+ redo of valid part of my previous revert.
-2006-04-09 Sebastien Moutte <sebastien@moutte.net>
+2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst/audioresample/debug.h:
- replace debug macros with variable number of parameters
- by a simple alias to gstreamer standard debug macros
- (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
- supported by MSVC 6.0 and 7.1)
- * gst/audioresample/resample.h:
- define M_PI and rint for WIN32
- * win32/common/libgstaudio.def:
- * win32/common/libgstriff.def:
- * win32/common/libgsttag.def:
- * win32/common/libgstvideo.def:
- add new exported functions
- * win32/vs6:
- update project files
+ * common:
+ * gst/audioconvert/gstchannelmix.c:
+ Revert "audioconvert: fixes warning: format not a string literal and no format arguments"
+ Revert this commit as unintentionally I've changed common.
+ This reverts commit 49ea0138223ec5f9e53780635cbcc70f33778667.
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
+2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.com>
- * ext/alsa/gstalsamixeroptions.c:
- (gst_alsa_mixer_options_class_init):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init):
- * gst-libs/gst/audio/gstaudiosrc.c:
- (gst_audioringbuffer_class_init):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
- * gst-libs/gst/interfaces/colorbalancechannel.c:
- (gst_color_balance_channel_class_init):
- * gst-libs/gst/interfaces/mixeroptions.c:
- (gst_mixer_options_class_init):
- * gst-libs/gst/interfaces/mixertrack.c:
- (gst_mixer_track_class_init):
- * gst-libs/gst/interfaces/tunerchannel.c:
- (gst_tuner_channel_class_init):
- * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- (gst_netbuffer_class_init):
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
- * sys/v4l/gstv4lcolorbalance.c:
- (gst_v4l_color_balance_channel_class_init):
- * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
- (gst_v4l_tuner_norm_class_init):
- * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
- Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
-
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfssrc: fixes warnings in macosx
+ warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'GnomeVFSFileOffset'
+ warning: format '%lld' expects type 'long long int', but argument 9 has type 'guint64'
- * ext/pango/gsttextrender.h:
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.h:
- * gst-libs/gst/video/gstvideofilter.h:
- * gst-libs/gst/video/gstvideosink.h:
- * gst/playback/gstplaybasebin.h:
- * gst/tcp/gstmultifdsink.h:
- * sys/v4l/gstv4lelement.h:
- Fix broken GObject macros
+2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
+ * gst/videorate/gstvideorate.c:
+ videorate: fix warning in macosx
- * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
- More debug to trace why my USB headset is not working with gst
+2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
-2006-04-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: fix warning in macosx
- * gst/playback/gstplaybasebin.c: (group_destroy):
- Clean up our group elements properly in the case where it never
- got committed - it still got added unconditionally to the bin.
+2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
-2006-04-07 Wim Taymans <wim@fluendo.com>
+ * common:
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
- * ext/theora/theoradec.c: (theora_dec_sink_event),
- (theora_handle_data_packet), (theora_dec_chain):
- Unref unhandled events.
- Protect against empty buffers.
- Perform QoS on running time.
+2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
-2006-04-07 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ audio: fix warnings building on macosx
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
- (gst_vorbis_enc_chain):
- Remove leaks from vorbisenc.
- Mostly minor changes, the only significant one is that now the
- buffers we set as 'streamheader' on the caps are copies of the
- original buffers, to avoid circular refcounting problems.
+2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-04-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: chwck formats just once per _chain()
- * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
- Don't remove our mute-probe if someone else already did so.
- Don't set a 2nd one if there is already one pending on the pad.
+2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
- (do_playbin_seek):
- When a seek fails, ensure that playbin is still set back to playing.
-
- * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
- (mpeg_ts_type_find), (plugin_init):
- Add a typefind function for mpeg-ts streams. (Closes: #336617)
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: add perf-log-category and log suboptimal operation
+ Log if we use an intermediate colorspace for conversion.
-2006-04-06 Andy Wingo <wingo@pobox.com>
+2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videorate/gstvideorate.c (gst_video_rate_reset)
- (gst_video_rate_init): Caps-related parameters should not be reset
- by a flush -- move their inits to the instance init function.
- (gst_video_rate_flush_prev): Don't complain if gst_pad_push
- is not OK, just return the result.
+ * common:
+ Automatic update of common submodule
+ From 19fa4f3 to a3e3ce4
- * gst/audiotestsrc/gstaudiotestsrc.c
- (gst_audio_test_src_class_init)
- (gst_audio_test_src_get_times): Re-enable is-live=true, as was
- broken by Stefan's commit on 24 March.
+2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
- buffers being pushed out. Fixes oggmux ! multifdsink.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Fix type-punning warning
-2006-04-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
- (gst_vorbis_dec_init), (vorbis_dec_finalize):
- * ext/vorbis/vorbisdec.h:
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
- (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
- (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
- (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
- (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
- (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
- (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
- (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
- (gst_vorbis_enc_buffer_from_packet),
- (gst_vorbis_enc_buffer_from_header_packet),
- (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
- (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
- (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
- (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
- (gst_vorbis_enc_change_state):
- * ext/vorbis/vorbisenc.h:
- Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
- vorbisenc adhere to the official nomenclature; use boilerplate
- macro.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Chains with an exposed endpad are complete too
+ This allows partial group changes, i.e. demuxer2 in the example below
+ goes EOS but has a next group and audio2 stays the same.
+ /-- >demuxer2---->video
+ demuxer--- \--->audio1
+ \--->audio2
-2006-04-04 Andy Wingo <wingo@pobox.com>
+2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
- Whoops, fix bug introduced. Bad hacker!
-
- * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
- Properly handle the case where you get EOS before any buffers are
- received. Use gst_buffer_make_metadata_writable where appropriate.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
- * ext/theora/theoradec.c (theora_handle_data_packet): This value
- is often negative -- make it signed so as not to wrap around.
- Fixes segfaults introduced on 9 March.
+2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-04-03 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Don't post missing plugin messages twice
+ decodebin2 already posts them after emitting the unknown-type signal,
+ there's no need to post another one.
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (theora_dec_src_event):
- Don't try to store a gdouble in a gboolean.
- Small cleanups.
+2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-04-03 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Rewrite autoplugging and how groups of pads are exposed
+ This now keeps track of everything that is going on, creates
+ a tree of chains and groups to allow "demuxer after demuxer" scenarios
+ and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
+ Also document everything in detail and give a general overview of what
+ decodebin2 is doing at the top of the sources.
+ Fixes bug #596183, #563828 and #591677.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
- Oggmux sucks.
- Make it suck slightly less by writing out the final page.
- Still can't encode a vorbis-in-ogg file correctly, though.
+2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-04-03 Andy Wingo <wingo@pobox.com>
+ * sys/ximage/ximagesink.c:
+ ximagesink: only start event thread if needed
+ The event thread is doing 20 wakeups per second to poll the events. If one
+ runs ximagesink with handle-events=false and handle-expose=false then we can
+ avoid the extra thread.
- * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
- a g_print.
+2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/theora/theora.c (plugin_init): Register theoraparse.
+ * ext/theora/theoraenc.c:
+ theoraenc: Make the default quality property 48.
+ This guarantees that people who use theoraenc without modifying any
+ properties will end up with a reasonably good quality output.
+ 48 is also the default of the encoder_example application shipped with
+ libtheora.
- * ext/theora/gsttheoraparse.h:
- * ext/theora/theoraparse.c: New files implementing a theora
- parser. Now we can properly remux ogg/theora+vorbis, yay.
+2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
- * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
+ * tests/check/libs/video.c:
+ tests/check/libs/video.c: Update strides for Y41B
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Fixed the sample pipeline (see #323798)
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: we can use GLib 2.18 API unconditionally now
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- * win32/common/config.h:
- * win32/common/config.h.in:
- use AS_VERSION and AS_NANO
- more cleanups
+ configure: bump GLib requirement to 2.18
+ Bump required GLib version as per the release planning docs.
-2006-03-31 Andy Wingo <wingo@pobox.com>
+2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
- uninitialized variable return that would happen.
-
- * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
- uninitialized variable return that would never happen.
-
- * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
- (vorbis_parse_sink_event): Add an event function to flush our
- state on a seek, and to drain buffers on a premature EOS.
- (vorbis_parse_push_headers, vorbis_parse_clear_queue)
- (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
- (vorbis_parse_chain, vorbis_parse_queue_buffer)
- (vorbis_parse_drain_queue): Queue up buffers until we can set
- their timestamps and granulepos values.
+ * gst-libs/gst/interfaces/tuner.c:
+ docs: clarify GstTuner docs in two places
- * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
- and keep track of data needed for deriving granulepos and
- timestamps for buffers.
+2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-03-29 Wim Taymans <wim@fluendo.com>
+ * sys/v4l/gstv4lelement.c:
+ v4l: fix compiler warning
+ Fix 'variable may be used uninitialized' compiler warning (which is
+ true in theory, but can't actually ever happen, since we always
+ call the function with check=FALSE).
+ Fixes #596313.
- Patch by: j^ <j at bootlab dot org>
+2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_class_init):
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c:
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/pango/gsttimeoverlay.c:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * gst/audioconvert/gstaudioconvert.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstogmparse.c:
* gst/subparse/gstsubparse.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- better/unified long descriptions
- Fixes #336477
+ * gst/subparse/mpl2parse.c:
+ * gst/subparse/tmplayerparse.c:
+ build: sprintf, sscanf need stdio.h
-2006-03-29 Wim Taymans <wim@fluendo.com>
+2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
- (stop_seek):
- Don't let double and triple clicks mess up our state.
-
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: only start event thread if needed
+ The event thread is doing 20 wakeups per second to poll the events. If one runs
+ xvimagesink with handle-events=false and handle-expose=false then we can avoid
+ the extra thread.
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_text_element), (gen_audio_element), (gen_vis_element):
- Error out gracefully when we can't create any of the usual
- conversion elements for some reason. Also, don't try to
- create an audioscale (sic) element that's not used anyway.
+2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/video/video.h:
+ Update Since tags for NV12/NV21
+ They are added in 0.10.26 now, not 0.10.25
- * gst/playback/gstplaybasebin.c: (setup_source):
- Don't post RESOURCE_NOT_FOUND error when we can't find a source
- element for a particular protocol, that's confusing for users.
- Instead, post a RESOURCE_FAILED error, so that our own error
- message is actually shown in totem etc. (#336303).
+2009-09-23 15:31:50 +0200 Benjamin Otte <otte@gnome.org>
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2009-09-23 11:03:57 +0200 Benjamin Otte <otte@gnome.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
- (gst_gnome_vfs_src_get_icy_metadata):
- Fix some minor memory leaks (#336194).
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-23 10:25:02 +0200 Benjamin Otte <otte@gnome.org>
- * ext/gnomevfs/gstgnomevfs.c:
- (gst_gnome_vfs_location_to_uri_string):
- * ext/gnomevfs/gstgnomevfs.h:
- * ext/gnomevfs/gstgnomevfssink.c:
- (gst_gnome_vfs_sink_set_property):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
- Make gnomevfssink accept filenames as well as URIs for the
- "location" property, just like gnomevfssrc does (and
- filesrc/filesink do) (#336190).
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Add NV12 and NV21 formats
-2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
- * tests/check/generic/clock-selection.c: (GST_START_TEST):
- set to NULL before unreffing, fixes a valgrind leak.
- Why was this not triggering the error that an object needs to
- be NULL before unreffing ?
- * win32/common/config.h:
- update
+ * gst-libs/gst/video/video.c:
+ [video] Fix Y41B
+ Chroma components should be aligned on 4byte boundaries.
+ https://bugzilla.gnome.org/show_bug.cgi?id=595849
-2006-03-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
- * gst/subparse/gstsubparse.c: (convert_encoding),
- (gst_sub_parse_change_state):
- * gst/subparse/gstsubparse.h:
- Text subtitle files may or may not be UTF-8. If it's not, we
- don't really want to see '?' characters in place of non-ASCII
- characters like accented characters. So let's assume the input
- is UTF-8 until we come across text that is clearly not. If it's
- not UTF-8, we don't really know what it is, so try the following:
- (a) see whether the GST_SUBTITLE_ENCODING environment variable
- is set; if not, check (b) if the current locale encoding is
- non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
- the current locale encoding is UTF-8 and the environment variable
- was not set to any particular encoding. Not perfect, but better
- than nothing (and better than before, I think) (fixes #172848).
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Fix Y41B
+ Chroma components should be aligned on 4byte boundaries.
+ https://bugzilla.gnome.org/show_bug.cgi?id=595849
-2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
- update core requirement to 0.10.4.1 because of async_playback
- vmethod on GstBaseSink
-
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/interfaces/streamvolume.c:
+ streamvolume: Define cbrt() if it's not available
+ Fixes build on Win32, bug #597537.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
- * gst/adder/gstadder.c: (gst_adder_init):
- use DEBUG_FUNCPTR for collectpads
+2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstfactorylists.c:
+ factorylist: Use gst_caps_can_intersect() instead of _intersect()
+ This is faster and results in less allocations.
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
- (gst_gnome_vfs_sink_class_init):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
- (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_base_init):
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_base_init):
- * gst/adder/gstadder.c: (gst_adder_get_type):
- * gst/adder/gstadder.h:
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
- (gst_audio_test_src_create):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_base_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
- * gst/volume/gstvolume.c:
- * sys/v4l/gstv4lmjpegsink.c:
- * sys/v4l/gstv4lmjpegsrc.c:
- * tests/check/libs/cddabasesrc.c:
- * tests/old/examples/gob/gst-identity2.gob:
- Add docs for adder, use GST_ELEMENT_DETAILS macro,
- define GstElementDetails at the top
-
-2006-03-23 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/common/libgstinterfaces.def:
- Add a lot of export functions for gst-python
- * win32/vs6/libgstinterfaces.dsp:
- Add a missing include folder in the project configuration
-
-2006-03-23 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time), (gst_base_audio_src_create),
- (gst_base_audio_src_change_state):
- Fix audio sources, forgot to make the ringbuffer
- startable...
+2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-23 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't set the external ghostpads blocked but only their targets
+ Pad blocks should never be done on external pads as outside elements
+ might want to use their own pad blocks on them and this will lead to
+ conflicts and deadlocks.
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time), (gst_base_audio_src_create),
- (gst_base_audio_src_change_state):
- unparent instead of unref the ringbuffer.
+2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-23 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Only use the object lock for protecting the subtitle elements
+ Using the decodebin lock will result in deadlocks if the subtitle encoding
+ is accessed from a pad-added handler.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
- (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
- Implement new async_play vmethod to start slaving and allow
- playback start in case of async PLAY state changes.
+2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- Enable QoS with new method in base class.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Improve debugging of pad blocks
-2006-03-23 Wim Taymans <wim@fluendo.com>
+2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Julien MOUTTE <julien at moutte dot net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
- (gst_video_test_src_do_seek), (gst_video_test_src_create):
- Partially handle 0 framerate, only EOS after the first frame
- is missing.
+2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
-2006-03-23 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ configure: Add an 'else' to pangocairo check
+ Otherwise it exits if it fails.
- Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c:
- Patch for support of YVU9 AVI files (#334822)
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: add pattern with out-of-gamut colors
+ Adds a pattern with out-of-gamut colors in a checkerboard
+ pattern with in-gamut neighbors. Useful for checking YCbCr->RGB
+ color matrixing. Correct matrixing and clamping will cause the
+ checkerboard pattern to be invisible.
-2006-03-22 Edward Hervey <edward@fluendo.com>
+2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/design/design-decodebin.txt:
- Added design document for new decodebin
- (Target Caps): text/x-pango-markup is also a default target caps.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: use CLOSE_SOCKET() instead of close()
+ Use CLOSE_SOCKET instead of directly calling close() because it does the right
+ thing for windows.
+ Fixes #597539
-2006-03-22 Wim Taymans <wim@fluendo.com>
+2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_dispose):
- Since we _parent the ringbuffer, we also need to
- _unparent instead of a plain _unref.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: fix printf variable type
+ Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
+ should be for guint64.
+ Fixes #596981
-2006-03-22 Wim Taymans <wim@fluendo.com>
+2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
- (stop_seek), (scrub_toggle_cb), (main):
- Add scrub checkbox.
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use the ffmpegcolorspace debug category
+ Move gstffmpegcodecmap debug to the ffmpegcolorspace category
-2006-03-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
- (gst_ogg_parse_chain):
- Fix very inefficient usage of linked lists (#335365).
+ * gst/gdp/gstgdppay.c:
+ gdppay: Don't repeat tags buffers for every new segment
+ Only send a tag buffer when one is received, not after every new segment
+ event/update.
-2006-03-21 Edward Hervey <edward@fluendo.com>
+2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
- * gst/playback/gstplaybin.c: (handoff):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_set_property):
- gcc 4.1 unreferenced pointer fixes.
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
- gst_buffer_ref() now takes a GstBuffer*.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: detect 'ftypqt ' as video/quicktime
-2006-03-20 Julien MOUTTE <julien@moutte.net>
+2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
- by Jan Schmidt.
+ * configure.ac:
+ back to development -> 0.10.25.1
-2006-03-19 Tim-Philipp Müller <tim at centricular dot net>
+=== release 0.10.25 ===
- * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
- (id3v1_type_find), (apetag_type_find), (plugin_init):
- Can't do tag preferences via probability, as tags would then
- lose against types that are recognised with MAXIMUM probability
- (like .wav); so let all tag typefinders return MAXIMUM themselves
- and order them via the rank. Split ID3v1 and ID3v2 typefinders so
- that we can prefer APE to ID3v1 (fixes #335028).
-
-2006-03-17 Wim Taymans <wim@fluendo.com>
+2009-10-05 13:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
- (gst_ring_buffer_may_start):
- * gst-libs/gst/audio/gstringbuffer.h:
- Only start playback if we are playing.
- should fix #330748.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ Release 0.10.25
-2006-03-17 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- * win32/common/config.h:
- Revert accidental commits to these files.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
- Patch by: Michal Benes <michal dot benes at xeris dot cz>
+ * ChangeLog:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.4 pre-release
- * tests/Makefile.am:
- Don't try to build tests in tests/icles if we
- don't have X (#323852)
+2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
- * gst-libs/gst/tag/gstid3tag.c:
- Add TXXX frame identifiers for replaygain stuff as used
- by some taggers (see #323721).
+2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-03-16 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstplaysink.c:
+ playsink: make the lock recursive for now
+ Fixes #583255
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_set_property),
- (gst_stream_selector_bufferalloc):
- Preserve the existing buggy streamselector behaviour by performing
- a fallback buffer allocation when downstream isn't linked yet.
- This should really be fixed in playbin by blocking pads until it's
- linked them.
- Also, use gst_pad_alloc_buffer instead of
- gst_pad_alloc_buffer_and_set.
+2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: fix the vis property getter
- * gst-libs/gst/tag/gstid3tag.c:
- Don't crash on unknown ID3v2 TXXX frames.
-
-2006-03-15 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
- Chain up to the parent finalize method.
- Add 32-bit sample size to the template caps.
+ * gst-plugins-base.spec.in:
+ Add missing file to spec file
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add the fourcc that the VMWare codec uses.
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_set_property),
- (gst_stream_selector_bufferalloc),
- (gst_stream_selector_request_new_pad):
- For the active pad, forward buffer-alloc requests, otherwise
- return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
- having to memcpy every frame when used by playbin.
+2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c:
- (gst_multi_fd_sink_handle_client_write):
- Get negotiated caps from the sink pad, rather than the sink
- pad's peer.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
-
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
- Don't forget to set src->callbacks_pushed to FALSE again when
- popping them, otherwise re-activation in a different mode won't
- work (#334620).
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: ignore URI fragments that look like device paths
+ Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
+ worked before the fix for bug #321532.
+ Also adds a check for negative track numbers and some unit tests for URI
+ parsing.
+ Fixes bug #595454.
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
- Patch by: Sebastien Moutte <sebastien moutte net>
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.3 pre-release
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
- (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
- (gst_ffmpeg_smpfmt_to_caps):
- Replace __VA_ARGS__ caps creation macros with varargs functions.
- Makes things compile on MSVC (#320765), looks nicer, and we can
- tell the compiler to check for the NULL terminator.
+2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: don't ever return NULL in list of strings.
- Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Make sure the buffer we copy into is really always big
- enough, this time for real (#333488).
-
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Expose mute,volume,vis-plugin and font-desc properties
+ https://bugzilla.gnome.org/show_bug.cgi?id=594623
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for 24bpp DIB (#305279).
+2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-03-14 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ GstPlaySink: Expose 'reconfigure' as an action signal.
- * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
- (gst_video_scale_init), (gst_video_scale_src_event):
- Re-enable QoS after the release.
- Rework videoscale to use the base class src_event handler.
+2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ GstPlaySink: Expose flags as a gobject property.
- * configure.ac:
- back to CVS.
+2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
-=== release 0.10.5 ===
+ * gst/playback/gstplayback.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playback: Register playsink as an element.
+ This allows using playsink from outside the playback plugin.
+ Add code to be able to request the sink pads using standard GStreamer API.
+ TODO : expose GObject properties/signals.
-2006-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- releasing 0.10.5, "Net"
+ * docs/libs/gst-plugins-base-libs.types:
+ docs: add new gst_stream_volume_get_type to types file
+ This is needs to get Gobject features to show up in the docs.
-2006-03-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
- * docs/plugins/Makefile.am:
- Part of previous cdparanoiasrc docs fixes, forgot to commit.
-
-2006-03-12 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix duration calculation for truncated files
+ If the last page of a stream has a granulepos of -1, that is,
+ it doesn't complete a packet, we need to continue to search
+ for the last granulepos.
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- Add cdparanoiasrc to docs.
+2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- More GstCddaBaseSrc docs.
+ * Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
+ This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
-2006-03-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
- * gst-libs/gst/tag/tag.h:
- API: libgsttag: new method gst_tag_from_id3_user_tag().
+ * ext/theora/theoraenc.c:
+ theoraenc: Fix a string leak in _getcaps()
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- NULL-terminate array of mpeg4 video file extensions.
- Fixes crash on PPC (#334226).
+ * ChangeLog:
+ * configure.ac:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.2 pre-release
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnome_vfs_src_check_get_range):
- gnome_vfs_uri_is_local() alone is not a good indicator
- whether we can operate in pull-mode with a specific URI,
- as it returns FALSE for file:// URIs that point to an
- NFS-mounted path. Be more conservative here: whitelist
- local files, blacklist http URIs and use the old
- mechanism for anything else (fixes #334216).
+ * tests/check/elements/audioresample.c:
+ check: Improve audioresample test
+ Make the audioresample test work with CK_FORK=no, and
+ turn a g_print into a GST_INFO.
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
- * configure.ac:
- back to trunk
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix crashes with even widths
+ The fix for green lines introduced by commit
+ 35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
+ for even widths. This patch fixes it.
-=== release 0.10.4 ===
+2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Implement GstStreamVolume interface
- * configure.ac:
- releasing 0.10.4, "Power"
+2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-10 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c:
+ volume: Implement GstStreamVolume interface
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- Disable max-lateness by setting it to -1 for now, so that
- we can bed QoS stuff in thoroughly between now and the next
- release.
+2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ * gst-libs/gst/interfaces/streamvolume.h:
+ * gst/playback/Makefile.am:
+ * win32/common/libgstinterfaces.def:
+ interfaces: API: Add GstStreamVolume interface
+ Fixes bug #567660.
- Patch by: Fabrizio Gennari
+2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Make sure we don't read beyond the palette buffer in case of
- broken or manipulated files (#333488)
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: properly fix the HTTP manual mode
+ When we're not parsing HTTP, return EPARSE when we get an HTTP
+ message.
-2006-03-10 Edward Hervey <edward@fluendo.com>
+2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix for variable not initialized.
+ * gst-libs/gst/interfaces/mixertrack.h:
+ mixertrack: add READONLY and WRITEONLY flags
+ Should really have been READABLE and WRITABLE, but those are hard to
+ add whilst maintaining backwards compatibility. See #343615.
+ API: GST_MIXER_TRACK_READONLY
+ API: GST_MIXER_TRACK_WRITEONLY
-2006-03-09 Wim Taymans <wim@fluendo.com>
+2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_get_type),
- (gst_visual_src_setcaps), (gst_vis_src_negotiate),
- (gst_visual_chain):
- Small cleanups.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: fix build against core that has debugging disabled
+ The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (gst_theora_dec_reset), (_theora_granule_time),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_header_packet), (theora_dec_push),
- (theora_handle_data_packet), (theora_dec_chain),
- (theora_dec_change_state):
- Add simple QoS.
+2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-09 Wim Taymans <wim@fluendo.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Add Since marker for the new skip-to-first property
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (audiocast_register_listener), (gst_gnome_vfs_src_start):
- Some cleanups.
+2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
-2006-03-09 Wim Taymans <wim@fluendo.com>
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ videorate: Make videorate work with a live source
+ Add a property that makes videorate skip to the first buffer it
+ receives instead of padding the stream from segment start to the
+ first real buffer.
+ Fixes bug #567928.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
- Don't try to activate NULL chains.
+2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/fft/gstfft.h:
+ * gst-libs/gst/fft/gstfftf32.h:
+ * gst-libs/gst/fft/gstfftf64.h:
+ * gst-libs/gst/fft/gstffts16.h:
+ * gst-libs/gst/fft/gstffts32.h:
+ fft: Mark one function as const and add notes that the structs should be private in 0.11
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix invalid memory access to region before peek'd data (#332964).
+2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-03-09 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add human readable format names when logging
+ Add string array with human readable names for format and type to be used in log
+ statements.
- Patch by: Christophe Fergeau
+2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
- * ext/pango/gsttextrender.c: (gst_text_render_init):
- * gst/adder/gstadder.c: (gst_adder_init):
- Don't leak padtemplates, closes #333510.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: don't print RTP timestamps as clocktime
+ Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
+ Fixes #594757
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
- Fix invalid memory access: make sure string passed to
- regexec() is NUL-termianted.
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ playbin(2): Document that the volume property uses a linear scale
+ Fixes bug #571610.
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mp3_type_find):
- Refactor mpeg/audio typefinding to make it more maintainable
- and easier to fine-tune. Make probing into middle of the file
- work properly (fixes #333900, also see #152688).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: don't return EPARSE
+ Don't blindly return EPARSE when http mode is disabled.
+ Restore old http mode after temporarily setting it to TRUE.
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c:
- (utf8_type_find_have_valid_utf8_at_offset):
- Remove part from previous commit that was bogus:
- g_utf8_validate() does in fact not accept embedded
- zeroes, so we don't need to check for those (thanks
- to Mike for the hint).
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: add ugly backward compat hack
+ Check for pulsesink < 0.10.17 because it includes code that is now included in
+ baseaudiosink. Disable that code in baseaudiosink to be compatible with the
+ older version.
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
- * gst/typefind/gsttypefindfunctions.c:
- (utf8_type_find_count_embedded_zeroes),
- (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
- Make plain/text typefinder more conservative: firstly, check
- for embedded zeroes, which are perfectly valid UTF-8 characters,
- but also a fairly good sign that something is not a plain text
- file; secondly, probe into the middle of the file if possible.
- If we can't probe into the middle, limit the probability value
- to be returned to TYPE_FIND_POSSIBLE (see #333900).
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
+ A green border could be visible when converting to Y444 or RGB, because
+ the last chroma samples weren't copied correctly
-2006-03-08 Michael Smith <msmith@fluendo.com>
+2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Make typefind function name for mpeg4 video unique.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix YVU9 and YUV9
+ - Buffer sizes were computed different from ffmpegcolorspace
+ - Green bar on right size for widths not divisable by 4
-2006-03-08 Wim Taymans <wim@fluendo.com>
+2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
- * ext/libvisual/visual.c: (gst_visual_init),
- (gst_visual_clear_actors), (gst_visual_dispose),
- (gst_visual_reset), (gst_visual_src_setcaps),
- (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
- (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
- (gst_visual_chain), (gst_visual_change_state):
- Cleanups, post nice errors.
- Handle sink and src events.
- Implement simple QoS.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix image for odd widths in some formats
+ videotestsrc rounds chroma down. This causes it to omit the last chroma
+ value completely for odd widths when the chroma is downsampled.
+ This patch special cases the last pixel to not be rounded down.
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- Use new basesink methods to configure max-lateness.
- Small doc update.
+2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
- Debug statement cleanups.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle kate and cmml as sparse streams too
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- Simple cleanup.
+2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: Better handling of sparse streams by sending segment updates
+ Fixes bug #397419.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
- (gst_text_overlay_init), (gst_text_overlay_set_property),
- (gst_text_overlay_get_property):
- Revert API/ABI break from March 1. Keep 'halign' and 'valign'
- as string type properties, but mark them deprecated. Add
- 'halignment' and 'valignment' properties that use enums
- instead of strings.
+2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ docs: tell a biit more about uri-decodebin and buffering
- Patch by: Fabrizio Gennari
+2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Allow palettes with less than 256 colours in AVI files
- (#333488)
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: take clock time in setcaps
+ Take the time of the clock so that the last_time field is set. This is important
+ for sinks that restart their internal ringbuffer after a caps change and need to
+ know the last know position.
-2006-03-07 Julien MOUTTE <julien@moutte.net>
+2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
- (gst_text_overlay_video_event): Fix wrong EOS handling on text
- pad. We were releasing the queued text buffer when we should keep
- it until video pad gets EOS or discard the text buffer because it's
- too old. That was eating the last subtitle buffer. Add some more
- debug.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: add some more debug
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
- (gst_text_overlay_video_chain):
- Fix invalid memory access (we can't access a buffer after it's been
- pushed downstream without taking a reference); fix memory leak (if
- there's no text to render, bail out before allocating stuff).
+ * ext/theora/theoraenc.c:
+ theoraenc: Print a debug message with supported formats
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
- (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
- * ext/pango/gsttextoverlay.h:
- If input is plain text, escape it before passing it to
- pango_layout_set_markup().
+ * ext/theora/theoraenc.c:
+ theora: Check supported input formats in getcaps function
+ We want to fail early when an older libtheora release is used that does
+ not support Y444 or Y42B formats, so use a getcaps function that does
+ this.
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
- Don't ignore flow return from gst_pad_push().
+ * ext/theora/theoraenc.c:
+ theora: Implement support in theoraenc for Y444 and Y42B
+ Fixes bug #594165.
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
- Patch by: Fabrizio Gennari
+ * ext/theora/theoraenc.c:
+ theora: Refactor the buffer copy code
- * ext/libvisual/visual.c: (gst_visual_getcaps),
- (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
- (gst_vorbisenc_convert_sink):
- * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
- (gst_audio_duration_from_pad_buffer):
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
- (gst_audio_filter_chain):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps):
- * gst-libs/gst/video/video.c: (gst_video_frame_rate),
- (gst_video_get_size):
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
- Don't leak references returned by gst_pad_get_parent()
- (#333663)
+2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
-2006-03-06 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoraenc.c:
+ theora: Split yuv_buffer creation into its own function
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- change location param details
- * gst/volume/gstvolume.c: (plugin_init):
- correct plugin description
+2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
-2006-03-06 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theora: Split out buffer resize in its own function
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
- (gst_gnome_vfs_src_check_get_range):
- Override GstBaseSrc::check_get_range() in order to avoid opening
- the resource just to check whether we can operate in pull-mode or
- not - we can predict that pretty well from the URI alone. Should
- fix problems with last.fm (#331690). (Requires latest core CVS).
+2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
-2006-03-06 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoraenc.c:
+ theora: Add assertions that functions don't fail
+ Some functions in libtheora can return an error, but that error cannot
+ ever happen inside theoraenc. In those cases assert that it doesn't.
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
- (gst_video_sink_class_init):
- Throw away frames that are later than 20 ms.
+2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-03-06 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: make stop state configurable
+ Make it easy to experiment with different stop states (NULL and READY)
- Patch by: Fabrizio Gennari
+2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Set depth on WMA caps (#333545)
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: correct for clock reset
+ When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
+ also make sure that the clock is updated with the elapsed time so that it
+ alsways increments even when the ringbuffer goes back to 0. When this happened
+ we need to adjust the sample position for the reset ringbuffer.
+ Fixes #594136
-2006-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
- put Theora BOS pages before others. This hardcodes
- the Ogg/Theora I profile, but hey.
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ baseaudiosink: whitespace fixes
-2006-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/README:
- updated with some examples
- * ext/theora/theoraenc.c: (granulepos_to_timestamp),
- (granulepos_add), (theora_buffer_from_packet):
- * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
- (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
- (gst_vorbisenc_chain):
- implement strategy from ext/ogg/README
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
- (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
- Fix muxer so that oggz-validate is happy with all streams;
- except for no eos mark, and the BOS page ordering
- * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
- (check_buffer_granulepos):
- * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
- update tests to check for OFFSET being set as requested
- fixed type of granulepos, it's not a ClockTime
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add more debug
-2006-03-05 Julien MOUTTE <julien@moutte.net>
+2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
- Check that the xvimage we are creating has a correct size
- before returning it. (#314897)
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ whitespace fixes
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Give id3 and ape tag typefinders a rank slightly higher
- than PRIMARY to ensure they're always run before any of
- the other typefinders (in particular wav and mp3) (#324186).
+ * gst-libs/gst/video/gstvideosink.c:
+ * gst-libs/gst/video/gstvideosink.h:
+ videosink: add "show-preroll-frame" property
+ Add a property to disable rendering of video frames during preroll. This
+ will only work for videosinks that use the new ::show_frame() vfunc instead
+ of overriding basesink's preroll and render vfuncs directly.
+ API: GstVideoSink:show-preroll-frame
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for '3IVD' fourcc (#333403).
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
-2006-03-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Bump requirements to GStreamer CVS for the new error enum.
+ * gst-libs/gst/video/gstvideosink.c:
+ * gst-libs/gst/video/gstvideosink.h:
+ video: add GstVideoSinkClass::show_frame()
+ Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
+ vfuncs and add some gtk-doc chunks.
+ API: GstVideoSinkClass::show_frame()
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
- Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
- space left on the device (fixes #333352).
+2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-03-04 Sebastien Moutte <sebastien@moutte.net>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: don't do stuff inside g_return_val_if_fail() statements
+ Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
- * win32/vs6:
- add a project file for libgstvolume
- update the workspace
+2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
-2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix compiler warning with MSVC
+ Fixes bug #594275.
- * ext/theora/theoraenc.c: (theora_set_header_on_caps):
- * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
- (GST_START_TEST):
- Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
- Set IN_CAPS on header buffers
+2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
-2006-03-02 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: fix event forwarding
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- Add audioresample to docs.
+2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
- * gst/audioconvert/gstaudioconvert.c:
- Add revision date.
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
+ Fixes #594258
- * gst/audioresample/gstaudioresample.c:
- (gst_audioresample_base_init), (gst_audioresample_class_init),
- (gst_audioresample_init), (gst_audioresample_dispose),
- (audioresample_get_unit_size), (audioresample_transform_caps),
- (resample_set_state_from_caps), (audioresample_transform_size),
- (audioresample_set_caps), (audioresample_event),
- (audioresample_do_output), (audioresample_transform),
- (audioresample_pushthrough), (gst_audioresample_set_property),
- (gst_audioresample_get_property), (plugin_init):
- * gst/audioresample/gstaudioresample.h:
- Added docs.
- Small code cleanups.
+2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-03-02 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ fix whitespace
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- Added videoscale to docs.
+2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
- * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
- (gst_video_rate_swap_prev), (gst_video_rate_event),
- (gst_video_rate_chain):
- Fix typo in docs.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: improve slave skew resync
+ The old one did the mistake of not actually advancing the ringbuffer, it just
+ adjusted the segbase, introducing the whole lenght of the ringbuffer as an
+ extra delay in the pipeline.
+ Also make sure that the resync can never go back in time, producing the same
+ timestamps that has already been produced, as this can cause severe problems
+ for sinks and other synching mechanisms.
+ Fixes #594256
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
- (gst_video_scale_init), (gst_video_scale_prepare_size),
- (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
- (gst_video_scale_fixate_caps), (gst_video_scale_transform):
- * gst/videoscale/gstvideoscale.h:
- Added docs, examples.
- Some code cleanups.
- Post errors instead of g_warning.
+2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-03-02 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: disable typefinder for headerless flac
+ Disable headerless flac typefinder as long as it happily typefinds anything
+ including /dev/urandom as flac and as long as it's not particularly useful
+ given that such streams don't really exist in the wild.
+ Also fix up some comments so that gtk-doc doesn't complain about them.
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- Added some more docs to libs and plugins.
+2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- Document ringbuffer some more.
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix small memory leak when setting window title
- * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
- (gst_video_rate_setcaps), (gst_video_rate_reset),
- (gst_video_rate_init), (gst_video_rate_flush_prev),
- (gst_video_rate_swap_prev), (gst_video_rate_event),
- (gst_video_rate_chain), (gst_video_rate_change_state):
- * gst/videorate/gstvideorate.h:
- Fix videorate to use segments.
- Make it work with 0/1 framerates (closes #331903)
- Handle EOS correctly.
- Added docs.
+2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
-2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix small memory leak when setting window title
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
- (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
- (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
- In state change function, first chain up to parent class,
- then handle downwards state change stuff. Remove some
- commented out cruft from 0.8 code.
+2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * .gitignore:
+ introspection: Add *.gir and *.typelib to .gitignore
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
- (gst_ogm_parse_chain):
- Don't remove/re-add source pad if the new caps are the same as
- the old caps anyway (#333042). When removing source pad, don't
- unref it afterwards - we didn't ref it when adding. Sprinkle some
- GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
- after using gst_pad_get_parent(). Return downstream flow return
- value in chain function.
+2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-03-02 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introduction: Fix out-of-tree build
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.signals:
- Fix hierarchy, added some more elements to the docs.
+2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_type):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
- Fix docs for ffmpegcolorspace.
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Fix introspection build by ordering sources/headers in dependency order
-2006-03-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
- (apetag_type_find), (ape_type_find), (plugin_init):
- Some typefinding fine-tuning:
- - rank ID3/APE tags in order of preference via probabilities, so that
- ID3v2 > APEv2 > APEv1 > ID3v1.
- - three or four bytes don't really justify MAXIMUM probability,
- change those to 'very likely' (musepack and monkeysaudio).
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Remove debug echo
-2006-03-01 Wim Taymans <wim@fluendo.com>
+2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/alsa/gstalsamixer.c:
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
- * ext/alsa/gstalsamixerelement.h:
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasink.h:
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
- (gst_alsasrc_init):
- * ext/alsa/gstalsasrc.h:
- Added alsa docs.
- Small code cleanups.
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix build of introspection data by using dependency order for the headers/sources
-2006-03-01 Wim Taymans <wim@fluendo.com>
+2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/Makefile.am:
- Dist new header too,
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Strip Gst prefix from all types/functions
-2006-03-01 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/gnomevfs/gstgnomevfssink.h:
- * ext/gnomevfs/gstgnomevfssrc.h:
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * ext/vorbis/vorbisdec.h:
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
- * ext/vorbis/vorbisenc.h:
- * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
- (vorbis_parse_chain), (vorbis_parse_change_state):
- * ext/vorbis/vorbisparse.h:
- * gst/audioconvert/gstaudioconvert.h:
- * gst/tcp/gsttcpserversink.h:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/volume/gstvolume.c:
- * gst/volume/gstvolume.h:
- Fix some more docs.
- Added docs for vorbisdec and vorbisparse.
- Fix vorbisparse.
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Fix build if gir-repository is not installed
-2006-03-01 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/pango/gstclockoverlay.h:
- * ext/pango/gsttextoverlay.h:
- * ext/pango/gsttextrender.h:
- * ext/pango/gsttimeoverlay.h:
- * ext/theora/gsttheoradec.h:
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * gst/audioconvert/gstaudioconvert.h:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- * gst/tcp/gstmultifdsink.h:
- Updated/added documentation.
+ * gst-libs/gst/video/Makefile.am:
+ video: Add gobject-introspection support
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
- (gst_text_overlay_halign_get_type),
- (gst_text_overlay_wrap_mode_get_type),
- (gst_text_overlay_base_init), (gst_text_overlay_class_init),
- (gst_text_overlay_init), (gst_text_overlay_set_property),
- (gst_text_overlay_get_property):
- Fix up properties to be enums instead of string to make bindings,
- introspection and automatic GUI creation possible.
- Add getters for the properties.
+2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-28 Sebastien Moutte <sebastien@moutte.net>
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Add gobject-introspection support
- * gst/audiotestsrc/gstaudiotestsrc.c:
- added defines of M_PI and M_PI_2
- * gst/ffmpegcolorspace/avcodec.h:
- removed #include "stdint.h" for win32 as _stdint.h is
- autogenerated to win32/common
- * win32/common/libgstaudio.def:
- * win32/common/libgsttag.def:
- added some exports
- * win32/vs6:
- some project files bugs corrected
- * win32/vs7:
- project files are reset to the default vs7 configuration
- (they link to msvcr71.dll using default optimizations)
-
-2006-02-28 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- Fix some docs.
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: Add gobject-introspection support
-2006-02-28 Edward Hervey <edward@fluendo.com>
+2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsasrc.c:
- Set proper class on the ElementDetails:
- Source/Audio instead of Src/Audio/
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ libs: Add nodist headers and sources to the introspection files
-2006-02-28 Edward Hervey <edward@fluendo.com>
+2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videoscale/vs_scanline.c:
- (vs_scanline_resample_nearest_RGBA):
- Revert optimization in videoscale. It should go in liboil and have
- an appropriate liboil function.
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Add gobject-introspection support
-2006-02-28 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock):
- Don't try to provide a clock in the NULL state.
+ * gst-libs/gst/rtp/Makefile.am:
+ rtp: Add gobject-introspection support
-2006-02-28 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
- (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
- (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
- Use GstSegment infrastructure to remove duplicated code
- and handle more seek cases correctly.
+ * gst-libs/gst/riff/Makefile.am:
+ riff: Add gobject-introspection support
-2006-02-28 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- Don't ignore return code from ffmpeg convert function.
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: Add gobject-introspection support
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
- Split out some long statements to ease debugging.
+2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/netbuffer/Makefile.am:
+ netbuffer: Add gobject-introspection support
- * ext/libvisual/visual.c: (gst_visual_init),
- (gst_vis_src_negotiate), (get_buffer), (plugin_init):
- Don't use gst_pad_use_fixed_caps, because it prevents downstream from
- being able to renegotiate the size. Instead, use the negotiation
- algorithm from the goom plugin to pick an initial output caps.
+2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Also, allow theoretical libvisual plugins that might support non-GL
- output even if they also do GL.
+ * gst-libs/gst/interfaces/Makefile.am:
+ interfaces: Add gobject-introspection support
-2006-02-26 Julien MOUTTE <julien@moutte.net>
+2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_init),
- (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
- (plugin_init): Load only non GL plugins. Fix some memleaks and
- possible negotiation issues.
+ * gst-libs/gst/fft/Makefile.am:
+ fft: Add gobject-introspection support
-2006-02-25 Julien MOUTTE <julien@moutte.net>
+2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
+ * gst-libs/gst/cdda/Makefile.am:
+ cdda: Add gobject-introspection support
+ This is disabled for now until gobject-introspection is fixed
-2006-02-24 Michael Smith <msmith@fluendo.com>
+2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
- (cmml_type_find), (plugin_init):
- Fix CMML type find function to not require a specific minor version
- of the CMML header.
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Add gobject-introspection support
- Add an MPEG4 video elementary stream typefind function.
+2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-04 Michael Smith <msmith@fluendo.com>
+ * configure.ac:
+ * gst-libs/gst/app/Makefile.am:
+ app: Add gobject-introspection support
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
- (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
- (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
- (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
- Annodex support in ogg demuxer. Doesn't do very much without the
- other annodex patches (to come).
+2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 00a859e to 19fa4f3
- Patch by: Fabrizio Gennari <fabrizio dot get at tiscali dot it>
+2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Pick up palette for MS video v1 (#327028)
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix midi typefinding
+ We already have a audio/midi typefinder so don't override it with the midi in
+ RIFF typefinder or else we fail to detect plain midi files.
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info),
- (gst_ffmpegcsp_get_unit_size):
- The 'palette_data' field from incoming RGB caps shouldn't be
- proxied on outgoing YUV caps; also, restrict unit size
- adjustment in case of paletted data only to the unit that
- actually has a palette. Fixes #330711.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: do buffering for more uris
+ Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
+ buffering.
+ Fixes #594020
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
- (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_get_unit_size):
- Plug some memory leaks.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinder for Midi inside RIFF
+ This is a standard Midi file format that should be supported by
+ all Midi decoders and also has the mimetype audio/mid according to
+ the Midi specification homepage.
+ Fixes bug #594094.
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/Makefile.am:
- * sys/xvimage/Makefile.am:
- Add some _CFLAGS and _LIBS that seem to be missing
- and/or required for Cygwin (see #317048).
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: add some debugging
-2006-02-22 Wim Taymans <wim@fluendo.com>
+2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsasrc.c:
- Fix description as pointed out by caugier.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: handle gaps
+ Add various conversion functions between time<->bytes<->rtptime that will be
+ used later on.
+ Refactor the min/max packet length code so that it can be used for both
+ sample/frame based payloaders. Cache the returned values.
+ code cleanups.
+ When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
+ same gap as the GStreamer timestamps gap.
-2006-02-22 Tommi Myöhänen <ext-tommi do myohanen at nokia dot com>
+2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Reviewed by : Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix frame duration calculations
+ Fix the calculation of the frame duration and rtp timestamps.
+ Add some debugging
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find):
- Better 3gp typefinding (#331526).
+2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppay: add some debugging
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
- Don't send EOS event here, the base class will send one for us.
+2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Subpictures without video stream aren't allowed either.
-
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
- Fix debug statement copy'n'paste-o.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: use offsets for RTP timestamps
+ Have a custom sample/frame function to generate an offset that the base class
+ will use for generating RTP timestamps. This results in perfect RTP timestamps
+ on the output buffers.
+ Refactor setting metadata on output buffers.
+ Add some more functionality to _flush().
+ Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
+ the next outgoing buffer.
+ Flush the pending data on EOS.
+
+2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: move function around
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
- Fix issues with mixer keeping state when muting/unmuting
- and when changing the volume whilst muted (see #331763
- and #331765).
+2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix sample duration calculation
- * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
- (parse_subrip), (gst_sub_parse_format_autodetect):
- Set right caps given that we send escaped text. Also,
- honour <i></i>, <b></b> and <u></u> markers that can be found
- in .srt files (fixes #310202).
+2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: more refactoring
+ Unify the sample/frame buffer handling code by making the functions plugable.
- * gst-libs/gst/audio/mixerutils.c:
- (element_factory_rank_compare_func):
- Make order in which elements are tried more determinable.
+2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-20 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ audiortppayload: refactor some more
+ Refactor getting the packet min/max size and alignment code.
+ Refactor converting bytes to time.
+ change some variable to something shorter.
- * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
- (remove_element_chain), (cleanup_decodebin),
- (gst_decode_bin_change_state): Make decodebin reusable by
- fixing remove_element_chain first and then introduce a
- cleaner in state change to ->NULL. (Closes #331678)
+2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-19 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ * win32/common/libgstrtp.def:
+ audiortppayload: refactor and cleanup
+ Always use the adapter when we need to fragment the incomming buffer. Use more
+ modern adapter functions to avoid malloc and memcpy. The overall result is that
+ the code looks cleaner while it should be equally fast and in some case avoid a
+ memcpy and malloc.
+ Use the adapter timestamping functions for more precise timestamps in case of
+ weird disconts.
+ Cache some values instead of recalculating them.
+ Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
+ the internal adapter.
+ API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
- use 0666 mask when creating files so umask gets applied
- correctly. Fixes #331295.
+2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Update common
- * gst/subparse/Makefile.am:
- * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
- (gst_ssa_parse_dispose), (gst_ssa_parse_init),
- (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
- (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
- (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
- (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
- * gst/subparse/gstssaparse.h:
- * gst/subparse/gstsubparse.c: (plugin_init):
- Add very basic parser for SSA subtitle streams (as often
- found in matroska files).
+2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: add property to disable perfect RTP time
+ Add a property to disable the generation of perfect RTP timestamps. By default
+ it is active.
+ API: GstBaseRTPPayload::perfect-rtptime
- * gst/playback/gstdecodebin.c: (mimetype_is_raw):
- That should be text/x-pango-markup, not text/x-pango-layout.
+2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-02-19 Julien MOUTTE <julien@moutte.net>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
- Polishing.
-
-2006-02-19 Julien MOUTTE <julien@moutte.net>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
- (gst_text_overlay_finalize), (gst_text_overlay_init),
- (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
- (gst_text_overlay_render_text), (gst_text_overlay_text_pad_link),
- (gst_text_overlay_text_event), (gst_text_overlay_video_event),
- (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
- (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
- Fix state change deadlock.
-
-2006-02-19 Julien MOUTTE <julien@moutte.net>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
- (gst_text_overlay_finalize), (gst_text_overlay_init),
- (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
- (gst_text_overlay_render_text), (gst_text_overlay_text_pad_link),
- (gst_text_overlay_text_event), (gst_text_overlay_video_event),
- (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
- (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
- * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
- and subtitles files.
-
-2006-02-19 Julien MOUTTE <julien@moutte.net>
-
- * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
- should be considered as raw.
-
-2006-02-19 Julien MOUTTE <julien@moutte.net>
-
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (cb_probe):
- * gst/playback/gststreaminfo.h: Introduce language informations.
-
-2006-02-18 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
- Set shared memory segments to be deleted as soon as we have attached,
- that way they get cleaned up automatically if we crash.
-
-2006-02-18 Julien MOUTTE <julien@moutte.net>
-
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
- * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
- functions are called with lock held.
-
-2006-02-18 Julien MOUTTE <julien@moutte.net>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
- (gst_text_overlay_finalize), (gst_text_overlay_init),
- (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
- (gst_text_overlay_render_text), (gst_text_overlay_text_pad_link),
- (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
- (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
- (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
- (gst_text_overlay_change_state): Refactoring of textoverlay
- without collectpads. This now supports sparse subtitles coming
- from a demuxer instead of a sub file. Seeking is still broken
- though. Need to discuss with wtay some more on how to handle
- seeking correctly.
- * ext/pango/gsttextoverlay.h:
- * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
- subtitles coming from the demuxer.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: allow subclasses to influence RTP time
+ Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
+ which RTP timestamps are generated. Usually timestamps are created from the
+ GStreamer timestamps on the buffer, which could result in imperfect RTP
+ timestamps.
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
- (gst_vorbisenc_convert_sink):
- Use some more scaling functions.
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppay: add macro to cast
-2006-02-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
- (gst_cd_paranoia_paranoia_callback),
- (gst_cd_paranoia_src_signal_is_being_watched),
- (gst_cd_paranoia_src_read_sector):
- * ext/cdparanoia/gstcdparanoiasrc.h:
- Add back 'transport-error' and 'uncorrected-error' signals and
- make them actually be fired when bad stuff happens (#319340).
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: code cleanups
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
- (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
- (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
- (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
- (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
- (gst_ring_buffer_pause), (gst_ring_buffer_stop),
- (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
- (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
- (gst_ring_buffer_clear):
- Small cleanups.
- Added some G_LIKELY.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppayload: don't check adapter
+ the adapter is never NULL so we don't need to check it.
+ Use _scale functions to avoid overflows.
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/TODO:
- Update TODO
+ * configure.ac:
+ * gst/typefind/Makefile.am:
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: move gio-based xdg mime typefinder from -bad to -base
+ Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
+ reporting a 20% probability and somesuch). Won't be registered if
+ the gio plugin has been disabled via ./configure --disable-gio.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_offset):
- When trying to play samples ASAP and we don't have a
- previous sample, try to play at position 0 instead of
- an invalid position.
+2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-17 Wim Taymans <wim@fluendo.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
- * ext/alsa/gstalsasink.c: (gst_alsasink_open),
- (gst_alsasink_reset):
- Also release lock when we get an error in _reset();
- fix an error message.
+2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: fix timestamping for when we do not have a clock yet
+ Should fix #559049.
- * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
- (gst_alsasink_init), (get_channel_free_structure),
- (caps_add_channel_configuration), (gst_alsasink_getcaps),
- (gst_alsasink_close):
- * ext/alsa/gstalsasink.h:
- Add support for more than 2 channels (#326720).
+2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: don't log not-yet-initialised integer value
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
- with 4 or 6 channels, assume a default channel layout to make things
- work (not sure there's anything else we can do in those cases).
+2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
+ And reflow code to be more indent friendly.
- * gst-libs/gst/audio/multichannel.c:
- Minor docs fix.
+2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
- Add support for WAVEFORMATEX, eg. PCM audio with more than two
- channels and a channel layout map.
-
-2006-02-16 Mathieu Garcia <b0nk at free dot fr>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: Make instance init faster by not reading /dev/urandom 3 times
+ ... which is the default seed when creating a new GRand. Because
+ GLib in older versions used buffered IO this would take a lot of time.
+ Instead use the global GRand for getting random numbers and keep the
+ three instance GRand for backward compatibility with a simple seed.
+ Fixes bug #593284.
- Reviewed by Edward Hervey <edward@fluendo.com>
+2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
- C-level optimization of the RGBA nearest neighbour function.
- Eventually this might end up in liboil with vectorized versions.
+ * gst/adder/gstadder.c:
+ adder: improve caps filter functionality. Fixes #590146.
+ Also use the capsfilter if there is no src-peer as the caps constrain what
+ we can do. Don't create any_caps as a default, as we check for NULL to skip the
+ filtering. This is a (small) performance regression as we always intersect
+ otherwise.
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_get_channel_positions):
- When we have more than 2 channels, but no channel layout is
- specified in the caps, return some default channel layout
- to the caller and warn about about a possibly buggy element
- (could be buggy filtercaps as well of course) (#317038).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post missing plugin messages before any error messages
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- Add gst-libs/gst/cdda to list of lib search paths.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: safely handle the indexes
-2006-02-15 Andy Wingo <wingo@pobox.com>
+2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
- timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
- to the Lord Jesus that I do not have to touch the ogg muxer ever
- again.
+ * win32/common/libgstrtsp.def:
+ def: add new rtsp symbols
-2006-02-15 Edward Hervey <edward@fluendo.com>
+2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- quicktime movie files can also contain 'uuid' atoms.
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: whitespace fixes.
-2006-02-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
- * gst/audioconvert/plugin.c: (plugin_init):
- Register the GstAudioChannelPosition enum type with the type
- system in the plugin_init function, so that it is known before
- any element actually makes use of multi-channel stuff. This is
- required for example if one wants to be able to deserialise/use
- a caps string with channel positions before any pipeline has
- been setup and started, like with gst-launch.
+ * gst/gdp/gstgdppay.c:
+ Bug 593035 - set IN_CAPS for streamheader buffer
-2006-02-14 Wim Taymans <wim@fluendo.com>
+2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
- (gst_ring_buffer_samples_done), (wait_segment),
- (gst_ring_buffer_commit), (gst_ring_buffer_clear):
- Add some compiler G_(UN_)LIKELY help.
- SIGNAL the ringbuffer waiters when going to PAUSED as well to
- make sure they can exit their functions. Should fix #330748
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: The internally linked pad of the selector might be NULL in some cases
-2006-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Makefile.am:
- * configure.ac:
- * win32/MANIFEST:
- * win32/common/_stdint.h:
- Windows does not have long long; copy the generated _stdint.h
- * win32/common/interfaces-enumtypes.c:
- (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
- (gst_mixer_track_flags_get_type),
- (gst_tuner_channel_flags_get_type):
- * win32/common/multichannel-enumtypes.c:
- (gst_audio_channel_position_get_type):
- update
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: Fix iterate internal linked pads functions for the stream selectors
+ This now used the new gst_iterator_new_single() function and as a side effect
+ fixes bug #592864.
-2006-02-13 Wim Taymans <wim@fluendo.com>
+2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Always sync on first sample we receive when starting.
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-read.c:
+ riff: Add support for AVF files
+ AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
+ Fixes bug #593117.
+
+2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-13 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Detect AVF files as RIFF files too
+ AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
+ Partially fixes bug #593117.
- * gst/playback/gstplaybin.c: (gen_vis_element):
- Update vis bin docs.
- Move queue after tee so we don't queue video buffers but
- audio samples instead. Fixes problems where the video queue
- is filled and the audio queue empty.
+2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-13 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/audioresample.c:
+ audioresample: Add unit test for checking for timestamp drifts
+ This also checks for perfect timestamping and offsetting.
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
- No need to push an EOS event here, GstBaseSrc will do that for us
- when we return FLOW_UNEXPECTED.
-
-2006-02-12 Wim Taymans <wim@fluendo.com>
+2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
- (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Use scale functions when possible.
- Fix error messages.
- Free clockid when after waiting for EOS.
- Use G_(UN_)LIKLY when it makes sense.
- Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix drain processing
+ In case we have to convert internally don't process output length input samples
+ but history length input samples.
-2006-02-12 Edward Hervey <edward@fluendo.com>
+2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Remove stray semi-colon (fixes #330888).
+ * tests/check/elements/audioresample.c:
+ audioresample: Improve debugging a bit in the unit test
-2006-02-12 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
- Fix up the XShm call testing so that we catch errors, and don't
- cause new ones by attempting to detach from a segment we failed
- to attach to. Fixes #312439.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: On the first buffer we need discont handling
+ Otherwise we won't get upstream timestamps and everything and all
+ output buffers would have -1 timestamps.
-2006-02-10 Edward Hervey <edward@fluendo.com>
+2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Added flv file typefind (video/x-flv).
+ * configure.ac:
+ * gst/subparse/gstsubparse.c:
+ subparse: Remove dependency on regex.h as it's not used anyway
+ Fixes bug #592544.
-2006-02-10 Edward Hervey <edward@fluendo.com>
+2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
- Also added the caps to the default set of riff video caps.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer overflow when pushing the drain
-2006-02-09 Andy Wingo <wingo@pobox.com>
+2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
- * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
- time and the end time of the last packet in the page.
- (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
- on the pages in our queue, set the duration as well. Reflow a
- debug statement.
- (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
- Fixes bad muxing order.
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: Fix timestamp drift
+ Fixes bug #591934.
-2006-02-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_setcaps), (gst_basertppayload_push):
- update seqnum before setting it on the packet; this makes sure
- that the timestamp and seqnum properties match after pushing
- a buffer
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/pango/gsttextrender.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ * gst/subparse/gstsubparse.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * sys/v4l/gstv4lsrc.c:
+ Remove Ronald Bultje from Authors field
+ Replaced with "GStreamer maintainers
+ <gstreamer-devel@lists.sourceforge.net>" or just removed,
+ depending on the number of other authors.
-2006-02-09 Andy Wingo <wingo@pobox.com>
+2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c
- (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
- overflow after 13.5 hours of recording. Kapow!
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix refcounting of _get_sink()
+ g_value_set_object() increases the refcount of the sink, which is not needed
+ because the object should already be refcounted. Make sure this is always the
+ case and use g_value_take_object().
+ Fixes: #592884
- * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
- the buffer size -- we don't care about underrun/overrun reporting
- right now, just need to return a useful value.
+2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Mark Transport as supporting multiple values.
- * configure.ac:
- Back to CVS
+2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-=== release 0.10.3 ===
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added missing Since tags.
-2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
- * configure.ac:
- releasing 0.10.3, "Under Pressure"
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: Improve audiosink startup performance
+ When we start the ringbuffer, immediatly continue processing samples if the
+ writer prepared some for us.
+ Fixes #545807
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
- * configure.ac:
- Drat. Bump libtool version number for new API.
- Prelease 0.10.2.3 (of 0.10.3)
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added new API for sending using GstRTSPWatch.
+ The new API to send messages using GstRTSPWatch will first try to send the
+ message immediately. Then, if that failed (or the message was not sent
+ fully), it will queue the remaining message for later delivery. This avoids
+ unnecessary context switches, and makes it possible to keep track of
+ whether the connection is blocked (the unblocking of the connection is
+ indicated by the reception of the message_sent signal).
+ This also deprecates the old API (gst_rtsp_watch_queue_data() and
+ gst_rtsp_watch_queue_message().)
+ API: gst_rtsp_watch_write_data()
+ API: gst_rtsp_watch_send_message()
+
+2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made gst_rtsp_watch_queue_data() thread safe.
- * configure.ac:
- * win32/common/config.h:
- 0.10.2.2 prerelease (of 0.10.3).
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_connection_set_http_mode().
+ With gst_rtsp_connection_set_http_mode() it is possible to tell the
+ connection whether to allow HTTP messages to be supported. By enabling HTTP
+ support the automatic HTTP tunnel support will also be disabled.
+ API: gst_rtsp_connection_set_http_mode()
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
- Revert Andy's newsegment change pending a more correct
- fix.
+2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
+ If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
+ then just setup the base64 decoding context for the first connection.
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find), (plugin_init):
- detect more files as 3gp
- group and reorder the iso file formats
+2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Write as much as possible in gst_rtsp_source_dispatch().
+ Try to write as much as possible if there are multiple messages queued.
- * ext/vorbis/vorbis.c: (plugin_init):
- Register musicbrainz tags, so apps don't have to.
+2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Add error_full callback to GstRTSPWatchFuncs.
+ The error_full callback is similar to the error callback, but allows for
+ better error handling. For read errors a partial message is provided to
+ help an RTSP server generate a more correct error response, and for write
+ errors the write queue id of the failed message is returned.
- * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
- (gst_tag_to_vorbis_tag):
- Make sure we called gst_tag_register_musicbrainz_tags()
- before possibly mapping a vorbiscomment string from/to a
- musicbrainz tag.
+2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made read_line() support LWS.
+ Rewrote read_line() to support LWS (Line White Space), the method used by
+ RTSP (and HTTP) to break long lines. Also added support for \r and \n as
+ line endings (in addition to the official \r\n).
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- In case we can't find the required number of consecutive
- mpeg audio frames to positively identify an MPEG audio
- stream, check if there's at least a valid mpeg audio
- frame right at offset 0 and if so suggest mpeg/audio
- caps with a very low probability (#153004).
+2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Do not split headers which should not be split.
+ From RFC 2068 section 4.2: "Multiple message-header fields with the same
+ field-name may be present in a message if and only if the entire
+ field-value for that header field is defined as a comma-separated list
+ [i.e., #(values)]." This means that we should not split other headers which
+ may contain a comma, e.g., Range and Date.
- * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
- a TIME segment if we get timestamped buffers. Requires recent
- fixes in core to work properly.
+2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Parse WWW-Authenticate headers correctly.
+ Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
+ allows commas both to separate between multiple challenges, and within the
+ challenges themself, we need to take some extra care to split these headers
+ correctly.
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Don't print the URI as part of the error message, it
- makes error dialogs look rather ugly, especially if
- the URI is very long or has characters in it that
- need escaping.
+2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improve parse_line().
+ Make parse_line() handle keys with multiple values on one line correctly.
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Error out if we have only text or subtitles, but nothing
- else. Also error out if we have subtitles but no video
- stream.
+2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Rewrote setup_tunneling().
+ Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
+ coded strings and duplicates of the message parsing code.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
- Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
- Post an error message on the bus when we encounter an
- error, which will hopefully be more meaningful than the
- 'Internal Flow Error' message users get to see if we
- just return GST_FLOW_ERROR.
+2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-07 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Rewrote gen_tunnel_reply().
+ Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
+ than a hard coded string.
- * configure.ac (GST_MAJORMINOR): Update core version req to
- 0.10.2.2, for the collectpads API addition (#330244).
+2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Ignore the Content-Length for POST requests.
+ The Content-Length for POST requests with an x-sessioncookie header should
+ be ignored as the length is bogus and only there to fool proxies.
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- Return FALSE from plugin_init() when GnomeVFS can't
- be initialised for some reason (#328423).
+2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-06 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Normalize lines (remove extra whitespace) before parsing.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
- Stick to seeking theory until i find the bug.
- * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
+2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made parse_string() return a result.
+ This will catch parsing errors when a too long string is received.
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_finalize), (theora_enc_sink_setcaps),
- (theora_set_header_on_caps), (theora_enc_chain),
- (theora_enc_change_state):
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- Make theoraenc and the tests leak free. Like, really.
+2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improved parsing of messages.
+ Do not abort message parsing as soon as there is an error. Instead parse
+ as much as possible to allow a server to return as meaningful an error as
+ possible.
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_finalize), (theora_enc_sink_setcaps):
- Add a finalize method to ensure we clean up state even if
- someone omitted the state change back to NULL.
+2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
- (gst_vorbisenc_chain):
- Free some more leaked bits.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added support for HTTP messages
- * tests/check/pipelines/theoraenc.c: (start_pipeline),
- (stop_pipeline):
- Wait for state changes to happen if they're ASYNC.
+2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- This ought to teach those fancy pants buildbots a lesson.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_connection_create_from_fd().
+ API: gst_rtsp_connection_create_from_fd()
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst-libs/gst/tag/gstid3tag.c:
- Add mapping for ID3 International Standard Recording Code
- tag "TSRC"
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Add initial buffer support.
+ The initial buffer contains data for a connection which should be used
+ before starting to actually read anything from the socket.
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
- Don't leak tag names.
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: don't block in paused
+ When we are asked to unlock we should either leave the render function or call
+ the wait_preroll method to release the stream lock.
+ Fixes #592657
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- * gst-libs/gst/tag/tags.c:
- Split libgsttag docs into multiple sections.
+ docs: fix includes for appsrc/appsink
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
- * gst-libs/gst/tag/gstvorbistag.c:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c:
- Add libgsttag to the docs.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Add support for the Authentication-Info header.
+ The Authentication-Info header is defined in RFC 2617 (Digest Access
+ Authentication).
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
- (gst_text_overlay_init), (gst_text_overlay_src_event),
- (gst_text_overlay_collected): Fix clockoverlay.
+ * ext/ogg/gstoggmux.c:
+ * tests/check/pipelines/oggmux.c:
+ oggmux: don't drop the streamheader field from the output caps
+ Revert previous 'fix' for bug #588717 and fix it properly, whilst
+ maintaining the streamheader field on the output caps. Also make
+ sure we don't leak header buffers we couldn't push when downstream
+ is unlinked. Add unit test for the presence of the streamheader
+ field on the output caps and for the issue from bug #588717.
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/compiling.sgml:
- Fix typo: it's pkg-config, not pkg-gconfig
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/tmpl/gstgconf.sgml:
- There is no libgstgconf in 0.10, remove it
- from the docs.
+2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Avoid duplicated headers.
+ Remove any existing Session and Date headers before adding new ones
+ when sending a request. This may happen if the user of this code reuses
+ a request (rtspsrc does this when resending after authorization fails).
- * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
- (gst_text_overlay_src_event), (gst_text_overlay_collected):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_class_init), (gst_sub_parse_init),
- (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
- (parse_mpsub), (parser_state_init), (handle_buffer),
- (gst_sub_parse_chain), (gst_sub_parse_sink_event), (plugin_init):
- * gst/subparse/gstsubparse.h: Introduce seeking code.
+2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Corrected the HTTP digest authorization computation.
+ Do not use sizeof() on an array passed as an argument to a function and
+ expect to get anything but the size of a pointer. As a result only the
+ first 4 (or 8) bytes of the response buffer were initialized to 0 in
+ auth_digest_compute_response() which caused it to return a string which
+ was not NUL-terminated...
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Add comment about LANGUAGE tag inconsistency (we want
- ISO-639-1, but extract three-letter identifiers?)
+2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/POTFILES.in:
- Add two translatable files.
+ * gst/playback/gstplaysink.c:
+ playsink: Also send SEEK events directly to a subpicture sink
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c:
- (gst_tag_register_musicbrainz_tags_internal),
- (gst_tag_register_musicbrainz_tags):
- Forward-port some tags stuff from the 0.8 branch. This is
- mostly the addition of musicbrainz tags and their mapping
- to vorbistags, and a vorbistag mapping of the language tag.
+ * gst/playback/gstplaysink.c:
+ playsink: If a custom text sink is used, send events to it too
+ Before, SEEK events would be sent to the video sink, which wouldn't
+ be linked in any way to the subtitle part of the pipeline and
+ subparse would never see the SEEK event. This would then seek
+ the audio/video but the subtitles would continue from the old
+ position instead.
+ Fixes bug #591664.
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
- refactoring.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Make missing plugins emit a warning message, not an error message
+ The problem with an error message is, that it will stop playback completely
+ while it could be that only a audio decoder plugin is missing and the video
+ could be played with the available plugins.
+ See bug #591677.
-2006-02-04 David Schleef <ds@schleef.org>
+2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggmux.c:
- * gst/typefind/gsttypefindfunctions.c:
- Add Dirac typefinding and add dirac format to oggmux.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post a correct error message for unknown types
+ Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
+ because a plugin is missing and nothing else is wrong.
+ Also make it an error instead of a warning.
+ Really fixes bug #591677.
-2006-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Don't put essential function call into
- g_return_*() macro, otherwise it'll all be
- replaced by NOOPs when compiling with
- G_DISABLE_CHECKS defined.
-
-2006-02-03 Edgard Lima <edgard.lima@indt.org.br>
-
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggparse.c:
- * gst/tcp/gsttcpserversink.c:
- * sys/v4l/v4lsrc_calls.c:
- * sys/v4l/v4lsrc_calls.h:
- Just make it compile with --disable-gst-debug.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post a missing plugin message additional to the error message on unknown types
+ Fixes bug #591677.
-2006-02-03 Wim Taymans <wim@fluendo.com>
+2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
- (gst_alsasink_class_init), (gst_alsasink_init),
- (gst_alsasink_write), (gst_alsasink_reset):
- * ext/alsa/gstalsasink.h:
- Add lock to protect alsa calls.
- Implement reset to flush samples ASAP, does not work
- with dmix though.
+ * gst/playback/gstplaysink.c:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ playbin2: fix error message string
+ Fixes #591577.
-2006-02-02 Wim Taymans <wim@fluendo.com>
+2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock):
- Ugh.. getting late I guess...
+ * gst-libs/gst/riff/riff-read.c:
+ riff: align API doc of gst_riff_parse_chunk with reality
-2006-02-02 Wim Taymans <wim@fluendo.com>
+2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock),
- (gst_base_audio_sink_set_property),
- (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
- Don't try to provide a clock when we are not negotiated since
- we might not be able to make it run.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid assertion failure on empty/NULL caps
-2006-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Unlinking two source pads is ... hard.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Also detect SVG by the <svg> starting tag
+ Not all SVG images have the DOCTYPE specified.
-2006-02-02 Wim Taymans <wim@fluendo.com>
+2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/TODO:
- Updated.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.18 function
+ g_checksum_reset() was added only in GLib 2.18, but we still require
+ only 2.16, so work around that if we only have 2.16. Fixes #591357.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
- On EOS, wait till the last sample is played before posting EOS.
+2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-01 Philippe Kalaf <burger at speedy dot org>
+ * tests/check/pipelines/streamheader.c:
+ streamheader: Fix caps leak in the vorbisenc unit test
- Patch by: Kai Vehmanen
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- Adds ability to enable newsegment bypass by setting queue_delay to
- zero. Also avoid thread being started if queue_delay is zero.
+2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/pipelines/streamheader.c:
+ checks: fix stream header unit test hanging in gst_task_cleanup_all()
+ Set pipelines to NULL state and unref when done.
- * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
- Make test work again by connecting fakesinks to each decoded pad,
- which makes the pipeline wait until each fakesink has a buffer
- queued before going to PAUSED state. At that point we know the
- decodebin pads are negotiated.
+2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/md5.c:
+ * gst-libs/gst/rtsp/md5.h:
+ rtsp: Use GLib's GChecksum instead of our own MD5 implementation
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
- (gst_cdda_base_src_handle_event):
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
- Pass unhandled queries to the parent class's query function.
+2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
- (gst_ogg_pad_src_query):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_query):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_query):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
- (gst_vorbisenc_sink_query):
- * gst/adder/gstadder.c: (gst_adder_query):
- Pass unhandled queries upstream instead of just
- dropping them (#326447). Also, fix supported
- query types list for some elements.
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix doc blurb typo for gst_navigation_send_key_event
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (au_type_find),
- (paris_type_find), (ilbc_type_find), (plugin_init):
- Fix typefinding for audio/x-au, audio/x-paris and
- audio/iLBC-sh. We cannot use the START_WITH macros
- here, because there can only be one typefind factory
- with the same name (caps), so the second one would
- replace the first one and the first one would never
- be called when doing typefinding (see #161712).
-
+ * gst/subparse/gstsubparse.c:
+ subparse: Allow . instead of , as millisecond delimiter in srt subtitles
+ Fixes bug #591207.
-2006-01-31 Wim Taymans <wim@fluendo.com>
+2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
- (vorbis_handle_header_packet), (vorbis_dec_push),
- (vorbis_handle_data_packet):
- Use scale_int when we can, add some more scaling.
- Check packettype before parsing it.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ Revert inlines that cause compiler warnings and are not needed anyway
-2006-01-31 Wim Taymans <wim@fluendo.com>
+2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/theora/theoradec.c: (_theora_granule_time),
- (theora_dec_src_convert), (theora_dec_sink_convert):
- Call right _scale functions.
- Use parameter instead of some other random value.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/riff/riff-media.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/video/gstvideofilter.c:
+ * gst-libs/gst/video/gstvideosink.c:
+ gst-libs: Remove dead assignments and resulting unused variables.
-2006-01-31 Wim Taymans <wim@fluendo.com>
+2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/theora/theoradec.c: (_theora_granule_frame),
- (_theora_granule_time), (_inc_granulepos),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_handle_type_packet), (theora_handle_data_packet),
- (theora_dec_chain):
- Use higher precision timestamps calculation.
- Convert some other conversions to _scale.
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gsttextrender.c:
+ * ext/vorbis/vorbisenc.c:
+ ext: Remove dead assignments and resulting unused variables.
-2006-01-31 Stefan Kost <ensonic@users.sf.net>
+2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_create_sine_table), (plugin_init):
- * gst/volume/gstvolume.c: (plugin_init):
- initialize gst_controller before using
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstfactorylists.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gststreamselector.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_image.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ gst: Remove dead assignments and resulting unused variables
-2006-01-31 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
- * tests/check/pipelines/theoraenc.c:
- * tests/check/pipelines/vorbisenc.c:
- Define constant using G_GINT64_CONSTANT to avoid errors when
- passing it around - otherwise it gets truncated to 32 bits.
+ * docs/design/draft-va.txt:
+ docs: add draft for generic introduction of video acceleration APIs idea
- Fixes failing tests.
+2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-01-31 Andy Wingo <wingo@pobox.com>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ Revert "theora: Convert theoradec to libtheora 1.0 API"
+ This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
+ Temporarily revert until we have a workaround for debian/ubuntu
+ packaging failure (see http://bugs.debian.org/528710).
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
- caps being set doesn't have a framerate value. Basically a stopgap
- measure.
+2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
- technically correct enough to put into core though.
- (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
- DURATION. Fixes theoraenc ! oggmux.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinders for many game sound console formats supported by gme
+ These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
- fraction, not double.
+2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-01-31 Sebastien Moutte <sebastien@moutte.net>
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix warning when we're not linked downstream and error out properly
+ Fix caps warning when there's no element linked downstream, and pass
+ not-linked flow return value correctly up the chain, so we error out
+ correctly. Fixes #588717.
- * win32/vs7:
- add vs7 project files created by Sergey Scobich
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
-2006-01-30 Sebastien Moutte <sebastien@moutte.net>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: Convert theoradec to libtheora 1.0 API
- * win32/vs8:
- add vs8 project files created by Sergey Scobich
-
-2006-01-30 Andy Wingo <wingo@pobox.com>
+2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
- timestamp + duration, not just timestamp -- ogg pages should be
- ordered by stop time. Necessary fix given the change in vorbis
- timestamps.
+ * ext/pango/gsttextrender.c:
+ textrender: Fix blitting of text over the output buffer and cairo painting
- * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
- (gst_theora_enc_init): Pull the granule shift out of the encoder.
- (granulepos_add): New function, handles the messiness of adjusting
- granulepos values.
- (theora_buffer_from_packet):
- (theora_enc_chain):
- (theora_enc_sink_event): Use granulepos_add, not +.
+2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/theoraenc.c
- (check_buffer_granulepos_from_starttime): Just check the frame
- count, not the actual granulepos -- we can't dictate to the
- encoder when it should be placing keyframes.
+ * ext/pango/gsttextrender.c:
+ textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
-2006-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
- SERVICE_NOT_AVAILABLE happens for example when you're trying to
- play an http:// stream from a server that's not serving
+ * tests/icles/test-colorkey.c:
+ colorkey-test: fix xsync error
-2006-01-30 Andy Wingo <wingo@pobox.com>
+2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
- * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
- * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
- remove the UINT64_CONSTANT macro, doesn't appear to be needed or
- available.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
- although theoraenc was timestamping correctly. Added handling of
- streams that start with nonzero timestamps.
+2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/Makefile.am:
- * tests/check/pipelines/theoraenc.c: New file, basically does same
- tests as vorbisenc.
+ * gst/playback/gstplaysink.c:
+ playbin2: smarter sink selection. Fixes #588523
+ Don't do fallbacks if application specified a sink element. When doing the
+ fallback use configured default elements instead of hardcoded linux only
+ elements. Improve error messages accordingly.
- * tests/check/pipelines/vorbisenc.c: I claim these bugs.
+2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2006-01-30 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstqueue2.c:
+ queue2: post error message when pausing task if so appropriate
+ If a downstream element returns an error while upstream has already
+ put all data into queue2 (including EOS), upstream will no longer
+ chain into queue2, so it is up to queue2 to perform some
+ EOS handling / message posting in such cases. See #589991.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
- (gst_audioringbuffer_pause):
- Implement pause that does not wait for completion.
+2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Don't drop buffers when going to PAUSED but perform preroll on
- remaining samples now that core base class supports this.
-
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
- (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
- (gst_ring_buffer_commit):
- Pause should not signal waiters.
- Implement return value of _commit correctly.
-
-2006-01-30 Andy Wingo <wingo@pobox.com>
-
- * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
-
- * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
- updated to timestamp from the first sample, not the last.
- (gst_vorbisenc_buffer_from_header_packet): New function, takes
- special care of granulepos and timestamp for header packets.
- (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
- when the first buffer has a nonzero timestamp.
-
- * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
- (GstVorbisEnc.subgranule_offset): New members. Take care of the
- case when the first audio buffer we get has a nonzero timestamp.
- (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
- properly timestamp vorbis buffers with the time of the first
- sample, not the last.
-
- * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
- vorbis_granule_time_copy -- now it takes the granule/subgranule
- offset into account.
-
- * tests/check/pipelines/vorbisenc.c: New test for correctness of
- timestamps, durations, and granulepos on buffers produced by
- vorbisenc.
-
-2006-01-30 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: change default slave method
+ Set the default slave method to the much better skew slaving algortihm.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
- Patch from Eric Jonas to support conversions to/from UYVY
- (Fixes: #324626)
+2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-01-30 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: make buffer writable
+ Make the input buffer writable before changing its contents.
- * gst/playback/gstplaybasebin.c: (group_commit), (queue_overrun),
- (setup_subtitle), (setup_source), (set_active_source):
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
- (gen_text_element), (gen_audio_element), (gen_vis_element),
- (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
+2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-01-29 Sebastien Moutte <sebastien@moutte.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix postscript typefinder probability
+ Two bytes for a rare format hardly warrants MAXIMUM typefinding
+ probability, POSSIBLE seems more appropriate.
- * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
- * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
- use gst_guint64_to_gdouble to be compliant with vs6
- * gst/playback/gstdecodebin.c: (try_to_link_1)
- * gst/videorate/videorate.c: (gst_video_rate_blank_data)
- use G_GINT64_CONSTANT for int64 constants
- * win32/common/libgstinterfaces.def:
- export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
- * win32/vs6:
- update and add new project files
-
-2006-01-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Makefile.am:
- * win32/MANIFEST:
- * win32/common/interfaces-enumtypes.c:
- (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
- (gst_mixer_track_flags_get_type),
- (gst_tuner_channel_flags_get_type):
- * win32/common/interfaces-enumtypes.h:
- * win32/common/multichannel-enumtypes.c:
- (gst_audio_channel_position_get_type):
- * win32/common/multichannel-enumtypes.h:
- add a win32-update rule like in core, and copy over enumtypes files
+ * ext/pango/gsttextoverlay.c:
+ pango: Send queries from the srcpad directly to the video sinkpad
-2006-01-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * win32/MANIFEST:
- * win32/common/config.h:
- * win32/common/config.h.in:
- add config files just like in core
+ * gst/subparse/gstsubparse.c:
+ subparse: Implement POSITION query
-2006-01-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
- Make gcc-4.1 happy (part of #327357).
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ subparse: Implement SEEKING query
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
- * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
- (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
- (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
- (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
- (gst_alsasrc_unprepare), (gst_alsasrc_read):
- Update all error messages. All of them should either use
- the default translated message, or actually provide a
- translatable string.
- Make the string for channel count problems meaningful.
+ * configure.ac:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
+ Require latest core for this.
+ Fixes bug #590430.
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- check for and throw RESOURCE_BUSY
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ pango: Add support for xRGB and BGRx formats
-2006-01-27 David Schleef <ds@schleef.org>
+2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
- checked in this change -- it requires liboil features not
- in 0.3.6. Revert parts.
+ * ext/pango/gsttextoverlay.c:
+ pango: Fix endianness issues from the pangocairo switch
+ cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
+ and BGRA on little endian architectures.
-2006-01-27 David Schleef <ds@schleef.org>
+2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * REQUIREMENTS:
- * configure.ac: update liboil requirement to 0.3.6
- * gst/videoscale/Makefile.am:
- * gst/videoscale/vs_scanline.c: liboilify
+ * ext/pango/gsttextoverlay.c:
+ pango: Re-add shading support which was dropped by a previous patch
-2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/libvisual/visual.c: (get_buffer):
- When pad_alloc returns a GstFlowReturn other
- than GST_FLOW_OK, make sure it is passed upstream.
+ * configure.ac:
+ * ext/pango/gsttextoverlay.c:
+ pango: Check if pangocairo supports vertical rendering and fix properties
-2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
- (gst_alsasink_class_init):
- Free the device name string.
+ * ext/pango/gsttextrender.c:
+ textrender: Use PROP_X instead of ARG_X consistently
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
- (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
- Don't remove a pad from the collectpads structure until it
- is released - it's a request pad, and may receive data again
- if the element gets moved back to PLAYING state.
+2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Ensure we turn on double buffering on the Xv port, and
- set the colour key to something dark and mysterious that
- isn't black.
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ pango: Some minor cleanup
-2006-01-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_base_init), (plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- - a library should not call setlocale. see "Libraries" node in
- gettext manual
- - make sure all plugins that use translation do bindtextdomain
- to point to the localedir
- * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
- (setup_sinks), (plugin_init):
- all this, and check for NULL when creating sinks
+ * configure.ac:
+ pango: Check for pangocairo instead of pangoft2
-2006-01-27 Julien MOUTTE <julien@moutte.net>
+2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
- (plugin_init): Make typefinding of subtitles work again.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ pango: Use pango-cairo instead of pango-ft2
+ pango-cairo will always use the native font rendering backend
+ of the platform and provides better results.
+ Fixes bug #340887.
-2006-01-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
- (mp3_type_frame_length_from_header), (mp3_type_find),
- (wavpack_type_find), (m4a_type_find), (ircam_type_find),
- (plugin_init):
- Backport a bunch of typefinding fixes from the 0.8 branch.
- Also, improve wavpack typefinding: if we can't peek the
- entire wavpack block, try to parse the bits we can get and
- see if we find what we're looking for in those.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add SVG typefinder
-2006-01-26 Julien MOUTTE <julien@moutte.net>
+2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c:
- (gst_ximagesink_calculate_pixel_aspect_ratio):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
- more cases of pixel aspect ratio.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add postscript typefinder
-2006-01-26 Edward Hervey <edward@fluendo.com>
+2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (pad_probe):
- Also consider the flush-start and tag events as unblockers
- for the pad probes.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Use static caps again for MPEG4 typefinding
-2006-01-26 Julien MOUTTE <julien@moutte.net>
+2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
- * gst/playback/gstplaybin.c: (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
- (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
- On the fly visualisation switch, works disabling, enabling as
- well but it won't be able to enable vis in a playbin that was
- created with no visualisation.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Implement better & more flexible MPEG4 typefinding
+ This detects more MPEG4 streams as MPEG4.
+ Fixes bug #556537.
-2006-01-25 Wim Taymans <wim@fluendo.com>
+2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Undo previous commit that returned WRONG_STATE sooner, it breaks
- resume after pause.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Allow to specify the device name in the URI
+ The allowed URI scheme is now:
+ cdda://(device#)?track
+ Also allow every combination of uppercase and lowercase
+ characters for the protocol part.
+ Fixes bug #321532.
-2006-01-25 Wim Taymans <wim@fluendo.com>
+2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
- (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
- Improve debugging.
- Post error when caps cannot be parsed.
- Resync on discontinuity in the stream.
- Clip samples to segment boundaries.
- return WRONG_STATE sooner when we are flushing.
-
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
- (gst_base_audio_src_get_time), (gst_base_audio_src_create):
- Make audiosrc operate in TIME.
- Set TIMESTAMP and DURATION on buffers.
-
-2006-01-24 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/examples/seek/seek.c: (main):
- Output tag messages as well.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Restrict width/height to 2^15 - 1
+ Otherwise integer overflows will happen, resulting in segmentation faults.
+ Fixes bug #590243.
-2006-01-23 Edward Hervey <edward@fluendo.com>
+2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (free_pad_probes), (remove_fakesink), (pad_probe),
- (close_pad_link), (gst_decode_bin_change_state):
- Replace GstPadBlockCallback with pad probes that detect
- first buffer AND eos before removing fakesink.
- Fixes hang with demuxers doing EOS while pre-rolling.
- Solves #328279
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix indention of template header
-2006-01-23 Andy Wingo <wingo@pobox.com>
+2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
- * ext/alsa/gstalsasink.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
-
- Patch by: Jens Granseuer <jensgr at gmx dot net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clarify documentation about caps and linkage
+ Fixes bug #589095.
-2006-01-22 Julien MOUTTE <julien@moutte.net>
+2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
- frames. We might get a frame destroyed after changing state to
- NULL, adding a safety check on xcontext.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix typefinding of SDP files
+ Fixes bug #589574.
-2006-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
- * gst-libs/gst/interfaces/xoverlay.c:
- Fix prepare-xwindow-id code example in the docs - we need to
- ignore all messages that aren't element messages as well.
-
-2006-01-21 Julien MOUTTE <julien@moutte.net>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Take the output offsets from the input if possible
+ Fixes bug #588915.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
- I think one day i'll completely undestand how caps negotiation
- is supposed to work. This refactoring handles buffer_alloc
- called with caps we can't handle. We definitely don't want a
- set_caps with those caps, so we define and allocate a buffer
- we would like to receive.
+2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Make sure to allocate enough memory for the temporary buffer
+ and fix scaling of odd-height interlaced video.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Free iterator when done.
+2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-01-17 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix interlaced scaling for I420
+ ...and some other minor mistakes in the previous change.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Fix playback of non-synchronised streams by assuming a rate
- of 1.0 instead of a random one.
+2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Makes this work again:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Include interlacing information in the AVPicture
+ This later allows to handle interlaced AVPicture different than
+ progressive ones which is needed for horizontally subsampled YUV
+ formats, see bug #589242.
- gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
- endianness=(int)4321, signed=(boolean)true, width=(int)16,
- depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
- audioresample ! alsasink
+2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.2 ===
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Add support for interlaced content
+ videoscale is not mixing content of two seperate fields anymore
+ and does scaling on every field separately.
+ Fixes bug #588761.
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
- releasing 0.10.2, "Then the devil is six"
-
-2006-01-16 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_set_property):
- Comment out broken code that connects to the state-changed signal.
- At this point, changing current stream selection is broken, but
- stuff like gst-launch playbin current-audio=1 works and filters
- to the chosen stream.
-
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix #327216 (null dereference in vorbisdec)
-
-2006-01-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/theora/theoradec.c: (theora_handle_comment_packet):
- Post taglist actually on bus instead of just freeing it
- (fixes #327114 and totem bug #327080).
-
- * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
- Use gst_element_found_tags_for_pad(), so that the tags
- are sent downstream as an event as well.
+ back to development -> 0.10.24.1
-2006-01-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
- (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
- (gst_xvimagesink_buffer_alloc):
- move all regularly occurring messages to GST_LOG level
- add some more object logs
-
-2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-plugins-base.doap:
+ Add 0.10.24 release to the doap file
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
- fix a silly segfault
+=== release 0.10.24 ===
-2006-01-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/mixerutils.c:
- * gst-libs/gst/audio/mixerutils.h:
- Add docs for mixerutils stuff.
-
-2006-01-13 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix playback for sources that emit raw audio or
- raw video streams (e.g.: cd audio sources) (#325984).
-
-2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/mixerutils.c:
- (gst_audio_mixer_filter_do_filter):
- actually save the element we create
-
-2006-01-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_handle_track_seek):
- No need to post a tag message on the bus when seeking
- within the same track, only post it when the current
- track changes.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Release 0.10.24
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (probe_triggered), (new_decoded_pad), (mute_group_type),
- (set_active_source):
- * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_base_init),
- (gst_stream_selector_set_property),
- (gst_stream_selector_request_new_pad):
- Reenable stream selection. These mechanisms need a complete overhaul
- in the face of 0.8->0.10 changes though.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggdemux.c:
- Change the pad template to src_%d to match the pads that
- are created from it. decodebin needs this information in order
- to decide that oggdemux is capable of producing multiple pads
- (and hence needs queues inserted).
+ * gst/typefind/gsttypefindfunctions.c:
+ * tests/check/gst/typefindfunctions.c:
+ typefinding: fix detection of fLaC id packet in broken flac-in-ogg
+ There are flac-in-ogg files without the usual flac packet framing
+ and these files just have a 4-byte fLaC ID packet as first packet.
+ We need to recognise the type just from these four bytes if we
+ want oggdemux to recognise these streams correctly.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected):
- Make debug output more useful by using GST_PTR_FORMAT.
+2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-01-11 Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ChangeLog:
+ * configure.ac:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.5 pre-release
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
- Set depth and width for alaw/mulaw (fixes #326601).
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: Don't assert on slightly different caps
+ Plugins should not assert on incompatible caps, caps negotiation will
+ fail anyway.
-2006-01-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/icles/Makefile.am:
- don't build the tests if we don't have the libs
+ * gst/adder/gstadder.c:
+ adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
- (gst_cd_paranoia_paranoia_callback):
- Don't try to free NULL pointers.
+ * configure.ac:
+ configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
+ The gio mount example needs GtkMountOperation, which is new in 2.14.
-2006-01-10 Edward Hervey <edward@fluendo.com>
+2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
- (gst_audio_rate_change_state), (plugin_init):
- Add debugging category.
- Fix type issues.
- Add case for incoming buffers without valid offset/offset_end.
+ * ext/alsa/gstalsasrc.c:
+ alsasrc: set alsasrc->handle back to NULL when closing device
+ Fixes crashes in gst_alsa_find_device_name() when probing or
+ reading the device-name property (e.g. when doing a dot-file
+ dump). Fixes #589797.
-2006-01-10 Michael Smith <msmith@fluendo.com>
+2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
- Don't leak GCond in audio sources.
+ * gst/playback/gststreamselector.c:
+ playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
+ Rename the GType of the pads of playbin's internal stream selector
+ element so they don't use the same type name as input-selector's
+ pads. Fixes #589622.
-2006-01-10 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstplaybin.c: (gen_audio_element):
- Don't leak an autoaudiosink/alsasink when we generate
- a new audio element. (old code, I guess)
+ * ChangeLog:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.23.4 pre-release
-2006-01-10 Michael Smith <msmith@fluendo.com>
+2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
- Support float audio in audiorate.
- Use width rather than depth for selecting sample width.
+ * tests/examples/v4l/.gitignore:
+ ignores: Ignore v4l probing example binary
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.h:
- Use GLib types here (that way we don't have to include the
- generated _stdint.h header, which makes life easier for win32
- folks that don't use autotools for the build) (#325990, patch
- by: Sergey Scobich).
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: recognise Kate spu subtitles as well
+ Recognise spu-subtitles, SUB and K-SPU as valid categories for
+ Kate subtitles as well.
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (wait_segment):
- * gst-libs/gst/audio/gstringbuffer.h:
- Name (private) union, makes Forte compiler happy (this time
- for real) (#324900).
+ * common:
+ Automatic update of common submodule
+ From fedaaee to 94f95e3
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
- * gst-libs/gst/audio/Makefile.am:
- Link against libgstinterfaces, needed for mixer
- and property probe stuff.
+ * gst-plugins-base.spec.in:
+ Update spec file with latest changes
-2006-01-09 Edward Hervey <edward@fluendo.com>
+2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/Makefile.am:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/audio-enumtypes.c:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/video-enumtypes.c:
+ 0.10.23.3 pre-release
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/mixerutils.c:
- (gst_audio_mixer_filter_do_filter),
- (gst_audio_mixer_filter_check_element),
- (gst_audio_mixer_filter_probe_feature),
- (element_factory_rank_compare_func),
- (gst_audio_default_registry_mixer_filter):
- * gst-libs/gst/audio/mixerutils.h:
- Add gst_audio_default_registry_mixer_filter() utility
- function.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: call send_event directly
+ We can't call gst_element_send_event() from a streaming thread as it gets the
+ state lock. Instead call the send_event method directly until we have a nice API
+ for this in basesrc.
+ Fixes #588746
-2006-01-03 Michael Smith <msmith@fluendo.com>
+2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/audioresample/resample.h:
- As before, but for o_buf
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiosink: Add stream-status messages
+ Fixes #587695
-2006-01-03 Michael Smith <msmith@fluendo.com>
+2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/audioresample/resample.h:
- Declare struct _ResampleState.buffer as unsigned char *, not void *,
- since we do arithmetic on it.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: Add stream-status messages
+ See #587695
-2006-01-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (wait_segment):
- * gst-libs/gst/audio/gstringbuffer.h:
- Sun's Forte compiler doesn't seem to like anonymous structs,
- so use same setup as in GstBaseSrc (fixes #324900).
+ * gst/adder/gstadder.c:
+ gstadder: Don't forget to free pending events on flush/dispose.
+ Fixes #588747
-2005-12-30 Stefan Kost <ensonic@users.sf.net>
+2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- * gst/volume/Makefile.am:
- * gst/volume/demo.c:
- move old example to tests/examples/volume/volune.c
- * tests/examples/Makefile.am:
- * tests/examples/seek/seek.c: (main):
- change window-close event from "delete-event" to "destroy"
- * tests/examples/volume/Makefile.am:
- * tests/examples/volume/volume.c: (value_changed_callback),
- (setup_gui), (message_received), (eos_message_received), (main):
- fix event handling and bus usage
+ * tests/check/elements/adder.c:
+ tests/adder: Add stream consistency checking. Fixes #588748
-2005-12-29 Stefan Kost <ensonic@users.sf.net>
+2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
- (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
- (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
- (gst_audio_test_src_create_square),
- (gst_audio_test_src_create_saw),
- (gst_audio_test_src_create_triangle),
- (gst_audio_test_src_create_silence),
- (gst_audio_test_src_create_white_noise),
- (gst_audio_test_src_create_pink_noise),
- (gst_audio_test_src_init_sine_table),
- (gst_audio_test_src_create_sine_table),
- (gst_audio_test_src_change_wave),
- (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
- (gst_audio_test_src_create), (gst_audio_test_src_set_property):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- update to basesrc changes, implement segmented seeking and eos
- handling, add a 'sine-tab' waveform for performance critical playback
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * po/POTFILES.in:
- ... and this time the other modified file that I missed last time.
-
-2005-12-29 Michael Smith <msmith@fluendo.com>
+ audiotestsrc: Make sure tags are properly serialized. Fixes #588746
+ We do this by letting the basesrc base class handle the tags.
- * gst/playback/gstdecodebin.c: (new_pad):
- Fix non-C89 variable declaration not at the start of a block. Should
- help some compilers.
+2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: Collect incoming tag events and send them after newsegment. Fixes #588747
- * tests/check/Makefile.am:
- And now fix 'make distcheck' (builddir != srcdir)
+2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Check for empty tag strings. Fixes #588724
- * configure.ac:
- * ext/cdparanoia/Makefile.am:
- * ext/cdparanoia/gstcdparanoia.c:
- * ext/cdparanoia/gstcdparanoia.h:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
- (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
- (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
- (gst_cd_paranoia_paranoia_callback),
- (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
- (gst_cd_paranoia_src_set_property),
- (gst_cd_paranoia_src_get_property), (plugin_init):
- * ext/cdparanoia/gstcdparanoiasrc.h:
- New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
- plugin again (there are still fixes required to playbin to make
- cdda:// uris work there).
+2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstqueue2.c:
+ queue2: fix leak and improve buffering
+ Keep track of the max requested position and compare this to the write position
+ in the temp file to get the current amount of buffered data.
+ Fix memleak of all incomming buffers.
+ Fixes #588551
- * tests/check/Makefile.am:
- Fix test case compilation.
+2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstinputselector.h:
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gstplaybin2.c:
+ playbin2: use private copy of input-selector
+ We shouldn't really depend on elements from -bad for stream
+ selection in playbin2, so use a private copy of input-selector
+ until the selector plugin is ready to be moved to -base or -good.
+ Fixes #586356.
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_update_duration),
- (gst_cdda_base_src_calculate_cddb_id):
- An integer is not a string. Fix access to uninitialised variable.
+2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Add cddabasesrc unit test; also actually enable the vorbis test.
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstinputselector.h:
+ playback: add private copy of the input-selector from gst-plugins-bad
+ Not hooked up yet though. See #586356.
- * tests/check/generic/states.c:
- Blacklist new cd audio elements as well.
+2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * tests/check/libs/cddabasesrc.c:
- Unit test for GstCddaBaseSrc (discid calculation mostly).
+ * tests/examples/v4l/Makefile.am:
+ examples: fix v4l probe example build
+ Fixes bug #588550.
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- Add docs for libgstcdda/GstCddaBaseSrc.
+ * ChangeLog:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.23.2 pre-release
- * gst-libs/gst/interfaces/mixertrack.h:
- Do one struct member per line with a semicolon at the end, that way
- even gtk-doc might parse it without complaining.
+2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+ * po/LINGUAS:
+ * po/tr.po:
+ Add Turkish translations
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/cdda/base64.c:
- * gst-libs/gst/cdda/base64.h:
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
- (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
- (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
- (gst_cdda_base_src_get_property),
- (gst_cdda_base_src_get_track_from_sector),
- (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
- (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
- (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
- (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
- (gst_cdda_base_src_uri_get_protocols),
- (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
- (gst_cdda_base_src_uri_handler_init),
- (gst_cdda_base_src_setup_interfaces),
- (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
- (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
- (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
- (gst_cdda_base_src_add_tags),
- (gst_cdda_base_src_add_index_associations),
- (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
- (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
- (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
- (gst_cdda_base_src_create):
- * gst-libs/gst/cdda/gstcddabasesrc.h:
- * gst-libs/gst/cdda/sha1.c:
- * gst-libs/gst/cdda/sha1.h:
- Add new libgstcdda with GstCddaBaseSrc class.
+2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/adder.c:
+ adder: One more attempt to fix the adder test
+ Give up and discard and recreate the alsasrc after checking it can
+ be opened, due to some strange crash inside alsa when we don't.
- * ext/gnomevfs/gstgnomevfssink.h:
- Use GstBaseSinkClass as parent_class member for class struct, not
- GstBaseSink.
+2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/adder.c:
+ adder: Perform get_state() in the unit test
+ Wait for the alsasrc to return to NULL after setting it to PAUSED for
+ testing, otherwise it leads to segfaults later on.
- Patch by: Michael Benes
+2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_class_init), (gst_video_test_src_start):
- Add start method to reset running time and number of frames sent
- when starting up (fixes #324696)
+ * tests/check/elements/adder.c:
+ adder: Don't fail when alsasrc is unavailable
+ Make the liveadder test succeed silently when it can't be completed
+ either because alsasrc is unavailable, or because the device is
+ inaccessible.
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.signals:
- Add docs stuff for gnomevfssrc and gnomevfssink.
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
+ Differentiate subtitle streams and lyrics/cracktastic/complex streams via
+ the category string in the headers. This seems like a useful distinction
+ to make, and also seems more future-proof. See #525743.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix example pipeline in gtk-doc blurb.
+2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggmux.c:
+ oggmux: add Kate caps to the list of accepted types
+ See #525743.
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
- (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
- (gst_gnome_vfs_handle_get_type), (plugin_init):
- * ext/gnomevfs/gstgnomevfs.h:
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
- (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
- (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
- (gst_gnome_vfs_sink_set_property),
- (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
- (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
- (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
- (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
- (gst_gnome_vfs_sink_uri_get_type),
- (gst_gnome_vfs_sink_uri_get_protocols),
- (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
- (gst_gnome_vfs_sink_uri_handler_init):
- * ext/gnomevfs/gstgnomevfssink.h:
- Port gnomevfssink; add gtk-doc blurb.
+2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
- (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
- (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
- (gst_gnome_vfs_src_uri_get_type),
- (gst_gnome_vfs_src_uri_get_protocols),
- (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
- (gst_gnome_vfs_src_uri_handler_init),
- (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
- (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
- (gst_gnome_vfs_src_send_additional_headers_callback),
- (gst_gnome_vfs_src_received_headers_callback),
- (gst_gnome_vfs_src_push_callbacks),
- (gst_gnome_vfs_src_pop_callbacks),
- (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
- (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
- (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
- * ext/gnomevfs/gstgnomevfssrc.h:
- s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
- file; add gtk-doc blurb with example pipelines.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: treat uri-schemas incasesensitive
+ Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
+ Fixes not showing buffering messages e.g. for HTTP://...
-=== release 0.10.1 ===
+2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
-2005-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: simplify docs
+ Make short-desc short - its used in the toc. Strip uneeded markup.
- * configure.ac:
- releasing 0.10.1, "Dobro Dedek"
+2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstvideo.def:
+ win32: Fix exports
+ Remove methods from video base classes that have moved to -bad.
+ Add gst_netaddress_to_string
-2005-12-21 Edgard Lima <edgard.lima@indt.org.br>
+2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/typefind/gsttypefindfunctions.c:
- iLBC30 and iLBC20 added to typefind.
+ * tests/examples/gio/.gitignore:
+ ignores: ignore the giosrc-mounting example binary
-2005-12-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init):
- update strings, values are in microseconds
- change the default sink buffer time to something that is smaller
- (to help software volume mixing have a slightly lower delay) but
- still be acceptable on Wim's laptop
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Add some partial documentation
+ Add a general documentation blurb for the GstNavigation functionality.
+ Still lacks some example code and detail on how to implement it.
+
+2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Siren codec and make two descriptions non-translatable
-2005-12-20 Edward Hervey <edward@fluendo.com>
+2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
- Made a quack, forgot to add DUCK to the riff video template.
+ * common:
+ Automatic update of common submodule
+ From 5845b63 to fedaaee
-2005-12-19 Edward Hervey <edward@fluendo.com>
+2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
- * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
- (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_chain):
- Make sure pads are initialized correctly.
* gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add a whole bunch of FOURCC <=> MimeType.
- Extend the riff video pad template to support the newly added fourcc.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add siren to the RIFF parser
+ Add siren7 caps to the RIFF parser.
-2005-12-18 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain):
- Extra debug output when activating/deactivating chains.
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/v4l/Makefile.am:
+ * tests/examples/v4l/probe.c:
+ v4lsrc: add a simple test case for device probing
- * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
- (is_demuxer_element), (try_to_link_1), (remove_element_chain),
- (unlinked):
- Remove a queue from our list when it becomes unlinked.
- Don't add queues to elements in class 'Demux' if they
- can only produce one pad
+2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
-2005-12-18 Julien MOUTTE <julien@moutte.net>
+ * configure.ac:
+ * sys/v4l/Makefile.am:
+ * sys/v4l/gstv4lelement.c:
+ v4lsrc: optional support for device probing with gudev
+ Enumerate v4l devices using gudev if available.
+ Fixes bug #583640.
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
- (gst_video_sink_get_type): Add a debug category.
+2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
-2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * gst/adder/gstadder.c:
+ adder: add since tags to docs
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
- Handle downstream newsegment by sending our own newsegment before the
- next buffer to be released. (#323900)
+2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * tests/examples/seek/seek.c:
+ seek: don't automatically start pipeline in DB
+ Keep the pipeline paused when we detect download buffering. The user has to
+ manually start the pipeline for now because we can't estimate when the buffering
+ will finish or when we have underrun.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- add queue delay to new segment as well (as opposed to just the first
- buffer). (bug #322347)
+2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-16 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstqueue2.c:
+ queue2: flush differently, avoiding deadlocks
+ Don't flush the file by closing and opening it but instead use g_freopen. This
+ avoids a deadlock in shutdown because we emit the temp-location property change
+ with the wrong lock held.
- * ext/libvisual/visual.c: (make_valid_name):
- change some char* into char[]
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
- (gst_audio_test_src_create):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- prepare to handle EOS and SEGMENT_DONE
+2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: add a checkbox for progressive download
- * tests/check/generic/states.c: (GST_START_TEST):
- Blacklist cdparanoia element in state test.
+2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Fix template construction
+ Fix the construction of the temporary filename construction as the application
+ name can be NULL and we don't want a separator between the prgname and the
+ template.
- Patch by: Benjamin Pineau
+2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- Add <string.h> includes for memset and FD_ZERO (fixes #323878)
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ playbin2: add support for progressive download
+ Add a new playbin2 flag (initially disabled) to enable progressive download
+ buffering in uridecodebin.
-2005-12-15 Michael Smith <msmith@fluendo.com>
+2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
- (gst_video_rate_chain):
- Fix timestamping for videorate when the first buffer it sees has a
- non-zero timestamp. Fix some misleading debug output.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add download property
+ Add a download property that will attempt to configure queue2 into progressive
+ download buffering.
+ Make sure we only enable download buffering for quicktime and flv formats.
-2005-12-15 Michael Smith <msmith@fluendo.com>
+2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c:
- Don't leak all input buffers to audioresample.
+ * gst/playback/gstqueue2.c:
+ queue2: add temp-template property
+ Add a new temp-template property so that queue2 can securely allocate a
+ temporary filename. Deprecate the temp-location property for setting the
+ location but still use it to notify the allocated temp file.
-2005-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
- Don't operate on empty text buffers. Strip newlines and
- tabs only from the end of the text, but leave them intact
- in the middle. Fix typo in gtk-doc description.
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: add a caps-property to avoid to need to plug a capsfilter afterwards
+ Adder can only handle one common format accross the pads. Thus one needed to add
+ a capsfilter afterwards and manage the caps. Now one can simply set the caps on
+ the property.
-2005-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybasebin.c:
- * gst/playback/gstplaybin.c: (handoff):
- Make sure the video frame buffer we return to apps via the
- "frame" property always has caps set on it. Modify
- _gst_gvalue_set_object() macro to handle NULL objects
- gracefully too.
+ * tests/check/elements/adder.c:
+ adder: skip live-seek text if we have no audiosrc, add new test
+ The seek-test needs a real audiosrc. Also add a test that checks that adder is
+ reusable. Finaly handle warnings as warnings to fix a assertion.
-2005-12-14 Stefan Kost <ensonic@users.sf.net>
+2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_class_init), (gst_audio_test_src_init),
- (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
- (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
- (gst_audio_test_src_create):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Adjust to some recent api changes and add wtays new cool seeking
- capabillities
+ * ext/gio/gstgiosink.c:
+ gio: Also post a "not-mounted" message from giosink
-2005-12-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsadeviceprobe.c:
- * ext/alsa/gstalsadeviceprobe.h:
- Helper functions to add device probing via the GstPropertyProbe
- interface to a class.
+ * tests/examples/gio/giosrc-mounting.c:
+ gio: Remove workaround for playbin2 bug in the sample application
+ The playbin2 bug was #588078.
- * ext/alsa/gstalsamixer.h:
- Comment out GST_ALSA_MIXER, it returns a struct that's not
- used.
+2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Add some debug info.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
+ If READY->PAUSED failed in the source element we would've swapped
+ the current and next group already. To allow READY->PAUSED to succeed
+ after the first failure we have to swap the current and next group
+ back again. This also ensure that we're again in the same state
+ as before the failed state change and not at the next group.
+ This was especially a problem for playbin2 pipelines that use the
+ new mounting support in giosrc as the source would fail for READY->PAUSED
+ the first time, the application mounts the location and then tries
+ to go READY->PAUSED again (and this time it would succeed).
+ Fixes bug #588078.
- * ext/alsa/gstalsamixerelement.c:
- (gst_alsa_mixer_element_interface_supported),
- (gst_implements_interface_init),
- (gst_alsa_mixer_element_init_interfaces),
- (gst_alsa_mixer_element_class_init),
- (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
- (gst_alsa_mixer_element_set_property),
- (gst_alsa_mixer_element_get_property),
- (gst_alsa_mixer_element_change_state):
- * ext/alsa/gstalsamixerelement.h:
- Add 'device' and 'device-name' properties. Add GstPropertyProbe
- for device handling (gnome-volume-control will need that).
+2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-12 Christian Schaller <uraeus@gnome.org>
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/gio/Makefile.am:
+ * tests/examples/gio/giosrc-mounting.c:
+ gio: Add example application that shows how to handle the "not-mounted" message
- * ext/Makefile.am: fix cdparanoia entry
- * gst-plugins-base.spec.in: add cdparanoia
+2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-12 Michael Smith <msmith@fluendo.com>
+ * configure.ac:
+ gio: Remove the experimental status from the GIO plugin
+ Fixes bug #510417.
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Use the correct function to free list of typefind factories.
+2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-12 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ gio: Add documentation for the new "not-mounted" and "file-exists" messages
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_class_init), (gst_video_test_src_init),
- (gst_video_test_src_parse_caps), (gst_video_test_src_query),
- (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
- (gst_video_test_src_create):
- * gst/videotestsrc/gstvideotestsrc.h:
- Implement seeking in videotestsrc.
- Small cleanups.
+2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-12 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgiobasesrc.c:
+ gio: Make sure that we have the correct stream position when starting
- * ext/cdparanoia/Makefile.am:
- * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
- (gst_paranoia_endian_get_type), (_do_init),
- (cdparanoia_class_init), (cdparanoia_init),
- (cdparanoia_set_property), (cdparanoia_get_property),
- (cdparanoia_do_seek), (cdparanoia_is_seekable),
- (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
- (cdparanoia_convert), (cdparanoia_get_query_types),
- (cdparanoia_query), (cdparanoia_set_index),
- (cdparanoia_uri_set_uri):
- * ext/cdparanoia/gstcdparanoia.h:
- Partially ported cdparanoia now that basesrc can support a
- plugin like this..
+2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-12 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgiobasesink.c:
+ gio: Make sure to flush the output stream if it shouldn't be closed
+ Otherwise there might still be unwritten data after the element
+ has stopped.
- * tests/examples/seek/scrubby.c: (main):
- Set higher priority for bus events so they don't get reordered with
- gtk gui events.
+2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
- (flush_toggle_cb), (main):
- Added checkbox to disable flushing seeks.
- Disable scrubbing when doing non flushing seeks.
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesink.h:
+ * ext/gio/gstgiobasesrc.c:
+ * ext/gio/gstgiobasesrc.h:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ gio: Don't close the GIO streams for the giostream{src,sink} elements
+ This makes it possible to do something useful with the streams
+ after the element has stopped. Fixes bug #587896.
-2005-12-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
- (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
- (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
- (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
- Implement some sort of event handling that doesn't rely on
- g_return_if_fail; make sure we always push the last chunk of an
- .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
- state change function; remove some old cruft. Seeking is still
- rather unlikely to work though.
+ * tests/check/pipelines/gio.c:
+ gio: Try to reuse the pipeline with the same stream objects
- * tools/.cvsignore:
- Ignore more.
+2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-11 Julien MOUTTE <julien@moutte.net>
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesrc.c:
+ gio: Improve the error message if a stream is already closed before usage
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- Fixed a leak of the current image reference when cleaning up.
- Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
+2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-09 Michael Smith <msmith@fluendo.com>
+ * ext/gio/gstgiosink.c:
+ gio: Post a custom file-exists message on the bus if the file already exists
+ An application can handle this message, remove the file in question
+ and restart the pipeline again without showing an error.
+ This fixes bug #529300.
- * tools/Makefile.am:
- * tools/gst-launch-ext-m.m:
- Remove gst-launch-ext. It doesn't work, and is no longer
- particularly useful.
+2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-08 Luca Ognibene <luogni@tin.it>
+ * ext/gio/gstgiosrc.c:
+ gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
- don't pass random values to ogmparse convert function.
- Make seeking possible in the exile1.ogm file.
+2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-07 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiosink.c:
+ gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Work around refcount problem with g_value_set_object() that occur
- if the core has been compiled against GLib-2.6 (g_value_set_object()
- will only g_object_ref() the element, but the caller will
- gst_object_unref() it and bad things will happen due to the way
- GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
- totem for people on FC4 using Thomas's 0.10 RPMs.
-
-2005-12-07 Edward Hervey <edward@fluendo.com>
+2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Time to welcome ogm to 0.10 :)
-
- * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
- (gst_ogg_pad_typefind):
- Oggdemux can now properly typefind elements with dynamic pads.
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Properly set caps on src pad, and set caps on outgoing buffers.
+ * ext/gio/gstgiosrc.c:
+ gio: Post a custom "not-mounted" message on the bus
+ This allows applications to mount the GFile if possible and restart
+ the pipeline instead of simply giving an error.
-2005-12-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixerelement.h:
- * ext/alsa/gstalsamixeroptions.h:
- * ext/alsa/gstalsamixertrack.h:
- * ext/alsa/gstalsasink.c:
- * ext/alsa/gstalsasink.h:
- * ext/alsa/gstalsasrc.c:
- * ext/alsa/gstalsasrc.h:
- * ext/cdparanoia/gstcdparanoia.h:
- * ext/gnomevfs/gstgnomevfsuri.h:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- * ext/pango/gsttextoverlay.h:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/vorbis/vorbisdec.h:
- * ext/vorbis/vorbisenc.c:
- * ext/vorbis/vorbisenc.h:
- * ext/vorbis/vorbisparse.h:
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiosink.c:
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstaudiosrc.c:
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/audio/multichannel.h:
- * gst-libs/gst/floatcast/floatcast.h:
- * gst-libs/gst/interfaces/colorbalance.c:
- * gst-libs/gst/interfaces/colorbalance.h:
- * gst-libs/gst/interfaces/colorbalancechannel.h:
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/mixertrack.h:
- * gst-libs/gst/interfaces/navigation.h:
- * gst-libs/gst/interfaces/propertyprobe.h:
- * gst-libs/gst/interfaces/tuner.h:
- * gst-libs/gst/interfaces/tunerchannel.h:
- * gst-libs/gst/interfaces/tunernorm.h:
- * gst-libs/gst/interfaces/xoverlay.h:
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/riff/riff-media.h:
- * gst-libs/gst/riff/riff-read.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.c:
- * gst-libs/gst/rtp/gstbasertppayload.h:
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- * gst-libs/gst/tag/gsttageditingprivate.h:
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_from_vorbiscomment_buffer):
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/video/video.h:
- * gst/adder/gstadder.c:
- * gst/adder/gstadder.h:
- * gst/audioconvert/audioconvert.c:
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.c:
* gst/audioconvert/gstchannelmix.c:
- * gst/audioconvert/gstchannelmix.h:
- * gst/audiorate/gstaudiorate.c:
- * gst/audioresample/buffer.h:
- * gst/audioresample/functable.h:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/resample.h:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
- * gst/ffmpegcolorspace/imgconvert.c:
- * gst/ffmpegcolorspace/imgconvert_template.h:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreaminfo.h:
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstfdset.h:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpplugin.h:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
+ audioconvert: Fix compilation when debugging is disabled
+ Fixes bug #587980.
+
+2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesink.h:
+ * ext/gio/gstgiobasesrc.h:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsink.h:
+ gio: Add vfunc for requesting the stream for the sinks too
+
+2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesink.h:
+ * ext/gio/gstgiobasesrc.c:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ gio: Some more random cleanup
+
+2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/gio/gstgio.c:
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesrc.c:
+ * ext/gio/gstgiobasesrc.h:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiosrc.h:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ * ext/gio/gstgiostreamsrc.h:
+ gio: Update my mail address and copyright
+
+2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/gio/gstgiobasesrc.c:
+ * ext/gio/gstgiobasesrc.h:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiostreamsrc.c:
+ * ext/gio/gstgiostreamsrc.h:
+ gio: General clean up and simplification
+ The GInputStreams are now requested by a vfunc from
+ the subclasses instead of relying that the subclass
+ sets it until it's needed.
+ This might also fix bug #587896.
+
+2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: keep sending newsegments after seeking
+ Adder sends with timestamps from 0 upwards. After seeking we need to send
+ new-segments to get correct positions-queries.
+
+2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/check/elements/adder.c:
+ adder: make test more robust
+ Add audioconverts to the live-seeking test to make it negotiate.
+
+2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: use core performance log category
+
+2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/adder/gstadder.c:
+ adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
+ This ensures that collectpads' cookie is properly updated so that when the streaming
+ threads will restart and be checking for the flushing status of all pads there will
+ be no inconsistent state.
+
+2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
+
+ * ext/pango/gstclockoverlay.c:
+ pango: Call tzset() before localtime_r()
+ POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
+ required to set the state variables that define the current timezone. Indeed,
+ glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
+ if the system timezone is changed for a running program between two calls to
+ gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
+ timezone equals /etc/localtime being modified.
+ Fixes bug #587676.
+
+2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
+
+ * ext/Makefile.am:
+ build: remove spurious schroedinger reference
+
+2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
+
+ * configure.ac:
+ * ext/Makefile.am:
+ * ext/schroedinger/Makefile.am:
+ * ext/schroedinger/gstschro.c:
+ * ext/schroedinger/gstschrodec.c:
+ * ext/schroedinger/gstschroenc.c:
+ * ext/schroedinger/gstschroparse.c:
+ * ext/schroedinger/gstschroutils.c:
+ * ext/schroedinger/gstschroutils.h:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstbasevideocodec.c:
+ * gst-libs/gst/video/gstbasevideocodec.h:
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideodecoder.h:
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ * gst-libs/gst/video/gstbasevideoencoder.h:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ * gst-libs/gst/video/gstbasevideoparse.h:
+ * gst-libs/gst/video/gstbasevideoutils.c:
+ * gst-libs/gst/video/gstbasevideoutils.h:
+ basevideo: send basevideo back to remedial school
+ Move basevideo classes and schroedinger plugin to -bad.
+
+2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ netaddress: add constant for max len
+
+2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ netbuffer: add gst_netaddress_to_string
+ Add function to serialize a net address to a string.
+ API: GstNetAddress::gst_netaddress_to_string()
+
+2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: make fd:// uri use buffering too
+ fd:// usually operate in push mode only and are thus suitable for buffering.
+
+2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/volume/gstvolume.c:
+ volume: include "1.0=100%" in property description
+
+2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstplaysink.c:
+ playsink: remove unused property defs
+
+2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/multichannel.c:
+ multichannel: rewrite the new doc comment a bit
+ Its part of the audio lib.
+
+2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Avoid a segfault when the video sink fails to start
+ Don't attempt to display the subpictures and segfault when the
+ video sink failed to start (and hence the videochain is NULL).
+
+2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ ringbuffer: add vmethod to clear the ringbuffer
+ Add a vmethod so that subclasses can be notified when they should clear the data
+ in the ringbuffer.
+
+2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: Fix the fourcc caps property for VC-1/WMVA
+ The caps property for carrying fourccs is 'format', not 'fourcc'
+
+2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: include in.h for FreeBSD compat
+ Fixes #586920
+
+2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * win32/common/libgstapp.def:
+ defs: add defs for new appsink buffer-list method
+
+2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ appsink: add docs and signals
+ Add docs for the new callback.
+ Add signals for the new buffer-list support.
+
+2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
+
+ * tests/check/elements/appsink.c:
+ Added unit tests for buffer list support in appsink.
+
+2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
+
+ * gst-libs/gst/app/gstappsink.c:
+ Added buffer list support.
+
+2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
+
+ * gst-libs/gst/app/gstappsink.h:
+ Added buffer list support.
+
+2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Include winsock2.h after defining WINVER.
+ Similar to bug #587080.
+
+2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Moved a comment.
+
+2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/multichannel.c:
+ docs: add basic section docs for multichannel and relocate the ones for audio
+ Add section docs for multichannel, so that it has a short desc in the toc too.
+ Move the section docs in adio up, so that the follow the copyright like
+ elsewhere.
+
+2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l/gstv4lelement.c:
+ * sys/v4l/gstv4lsrc.c:
+ v4l: open/close device in ready.
+ Simillar change like in v4l2src. This allows probing feature in paused, where
+ streaming is noit yet started.
+
+2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: fix initial volume handling also when reusing the element
+ This is a follow-up to commit 452988, making it work correctly when the audio
+ chain is reused.
+
+2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Define WINVER before including any win headers
+ Fixes bug #587080.
+
+2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
+
+ * gst-libs/gst/riff/riff-read.c:
+ riff: prevent crash if rounded up tag size exceeds data size
+ When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
+ and an invalid read past the buffer data follows.
+
+2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/video/gstbasevideocodec.c:
+ basevideocodec: By default don't allow caps changes on the srcpad
+ This fixed playback of Dirac files with schrodec when upstream wants
+ a different width/height, basevideocodec accepts this and then
+ pushes buffers with new caps but content of the old caps.
+ In the best case this will just result in wrong unit size and a
+ failure in basestransform elements.
+
+2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * autogen.sh:
+ autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
+ Check for more automake command variants. Use printf instead of 'echo -n'
+ for portability
+
+2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ Automatic update of common submodule
+ From f810030 to 5845b63
+
+2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstscreenshot.c:
+ screenshot: don't leak message
+
+2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
* gst/typefind/gsttypefindfunctions.c:
- * gst/videorate/gstvideorate.c:
- * gst/videotestsrc/gstvideotestsrc.h:
+ typefinding: lower the h264 typefinder's probability
+ A NEARLY_CERTAIN is absolutely not warranted given the kind
+ of things it checks for. Even a LIKELY is probably not entirely
+ appropriate.
+
+2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
+
+ * common:
+ Automatic update of common submodule
+ From f3bb51b to f810030
+
+2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for multipart
+ So we get slightly nicer error messages when multipartdemux is missing.
+
+2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/adder/gstadder.c:
+ adder: only unflush when we flushed before
+ Ass suggested by Stefan Kost:
+ Keep track of when the sinkpad was set to flushing and unflush the pad when an
+ upstream flushing seek failed.
+
+2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix leak when the source fails to change state
+
+2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/subparse/gstssaparse.c:
+ ssaparse: avoid leaking all buffers
+
+2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/check/elements/adder.c:
+ adder: test seek handling in adder
+ This tests seeking on an adder that has a normal and a live source connected.
+ Wheter the current behavior is the desired one needs to be discussed still
+ (see #586033)
+
+2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: pass the xwindow along to not look at the yet unset var.
+ When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
+
+2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/ximage/ximagesink.c:
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ x(v)imagesink: catch tags and show title in own window
+ Refactor the code that sets the window title. Catch tag-events and use title
+ metadata for the window title.
+
+2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
+ Also make all the function arrays constant.
+
+2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: Add support for generating gaussian white noise
+ This patch adds support for stationary white Gaussian noise.
+ The Box-Muller algorithm is used to generate pairs of independent
+ normally-distributed random numbers.
+ Fixes bug #586519.
+
+2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix NV12 and NV21 transformations
+ Fix some stride problems, fix the nv12 to nv21 direct transformation,
+ and implement a direct conversion to yuv444 to save CPU.
+
+2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix NV12 painting for odd strides/heights
+
+2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
+ cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
+ Finally fixes #531035.
+
+2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ cdparanoia: try to guess a good cache size if it's set to -1
+ Try to guess from the paranoia-mode setting whether playback or
+ ripping is wanted, and use a smaller cache size if we're likely
+ to be doing playback, to avoid a long startup delay. Since this
+ was the value used in older cdparanoia versions, it should be
+ fine in any case. See #586331.
+
+2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
+
+ * configure.ac:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ cdparanoia: expose cache size setting
+ This setting was added in cdparanoia 10.2. The default value is good
+ for audio extraction, but lower values (previous versions of cdparanoia
+ used 150) are better for realtime playback.
+ Fixes #586331.
+
+2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
+
+ * gst-plugins-base.spec.in:
+ Make build of schro plugin conditional
+
+2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ * win32/common/libgstrtp.def:
+ basertppayload: add support for bufferlists
+ Based on patch from Ognyan Tonchev.
+ See #585559
+
+2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtpbuffer: use new convenience functions
+ New core convenience functions makes the list getters and setters trivial.
+ Maybe even too trivial...
+
+2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * win32/common/libgstrtp.def:
+ defs: add new symbol to win32 defs file
+ Based on patches by Ognyan Tonchev.
+ See #585559
+
+2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: cleanups, add _list_get_seq() too
+ Clean up the docs a little.
+ Add missing _list_get_seq method.
+ Add new symbols to the docs
+
+2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * win32/common/libgstrtp.def:
+ rtp: cleanups
+ Add Since tags to docs
+ Move some code around
+ Add win32 symbols
+
+2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtp: add bufferlist support
+
+2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: pass data to macros instead of GstBuffer
+
+2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_watch_queue_data() to the exports
+ Fix the tests by exporting the new symbol from the win32 dlls
+
+2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: appname might be NULL
+ Don't set title if appname is unknown.
+
+2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: set window title from application name
+
+2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Made the parsing of the RTSP URL scheme more generic.
+
+2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_watch_queue_data().
+ gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
+ but allows for queuing any data block for writing (much like
+ gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
+ API: gst_rtsp_watch_queue_data()
+
+2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Only extract the session ID from RTSP responses.
+
+2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
+
+2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
+
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improved base64 decoding in fill_bytes().
+ The base64 decoding in fill_bytes() expected the size of the read data to
+ be evenly divisible by four (which is true for the base64 encoded data
+ itself). This did not, however, take whitespace (especially line breaks)
+ into account and would fail the decoding if any whitespace was present.
+
+2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audiosrc: fix get_offset
+ When we need to jump to the most recently captured sample, jump to where the
+ next sample will be written instead of to some old data.
+ Fixes #581460
+
+2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: free the ringbuffer when going to NULL
+ Unparent and free the ringbuffer when going to NULL, like we do with the
+ audiosrc element. We can do this now because we correctly manage the time
+ jumping back to 0.
+
+2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audio: correctly handle short read/writes
+
+2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: add some extra logging for buffer timestamps
+
+2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/adder/gstadder.c:
+ adder: more seeking fixes.
+ When a seek failed upstream, make sure the adder sinkpad is set unflushing again
+ so that streaming can continue.
+ We only have a pending segment when we flushed.
+ Set the flush_stop_pending flag inside the appropriate locks and before we
+ attempt to perform the upstream seek.
+ Add some more comments.
+ Use the right lock to protect the flags in flush_stop.
+ See #585708
+
+2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Free iterator after removing all groups
+
+2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideofilter.c:
+ videofilter: Add a default get_unit_size function
+ This returns the correct values for all formats that are handled by
+ GstVideoFormat and makes all the custom get_unit_size functions in
+ many elements unnecessary.
+
+2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add Timestamp header field
+ fixes #585994
+
+2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: set smarter target state on uridecodebin
+ Set the target state of the newly added uridecodebins to somthing else that
+ PAUSED so that we keep their state in sync with the playsink state.
+ Fixes #585268
+
+2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: set the sink flag on the element
+
+2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add debug message
+
+2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosink, audiosrc: do the class_ref()s in the right class_init functions
+ Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
+
+2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosink,audiosrc: ref the audio ring buffer class and type in class_init
+ Hack around thread-safety issues in GObject and our racy _get_type()
+ functions (we could easily fix the _get_type() functions, but we still
+ need to hack around the GObject class races until we require a newer
+ GLib version, I think).
+
+2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audiosrc: return FALSE when receiving a SEEK event
+ When receiving a seek event, return FALSE as we don't implement seeking.
+
+2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/examples/seek/seek.c:
+ Don't use deprecated GTK API
+ Fixes bug #585758.
+
+2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: send flush_stop when seeking failed
+ At least do the fix to sent the flush_stop when seeking failed to ensure we
+ keep no pads flushing. before it was send when the seeking worked which is just
+ plain wrong and was not the intention.
+
+2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use a more consistent naming of GstRTSPRec variables.
+
+2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Call message_sent() callback for all sent messages.
+ Previously the messages_sent() callback was only called for messages
+ which had a CSeq, which excluded all data messages. Instead of using the
+ CSeq as ID, use a simple index counter.
+
+2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/theora/theoradec.c:
+ * ext/vorbis/vorbisdec.c:
+ oggdemux: post/send tags with the container-format tag
+ For this to work properly, theoradec and vorbisdec need to put
+ tag events received from upstream into the pending_events list
+ so they get pushed out after any newsegment event, not before.
+
+2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ * tests/old/examples/seek/cdplayer.c:
+ Don't use deprecated GTK API
+ Fixes bug #585758.
+
+2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/adder/gstadder.c:
+ adder: send flush-stop earlier
+ When no flush-stop has been sent by upstream, we have to send one ourselves to
+ continue playback. Do this as soon as the collect function is called instead of
+ after we possibly pushed segment events (that got then flushed out)
+
+2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/seek.c:
+ seek: add shuttle controls
+
+2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/stepping2.c:
+ example: fix compile
+
+2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/Makefile.am:
+ examples: build the stepping2 example
+
+2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: update for new step API
+
+2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: do reverse seeks more accurate
+ For reverse seeking with the accurate flag set, try to be more precise by
+ seeking a little bit after the requested position.
+
+2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstogmparse.c:
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstssaparse.h:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
+ Make subtitle parsers post a taglist with codec tags, so the application
+ knows what kind of subtitle a subtitle stream is. Fixes #576552.
+
+2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: handle border cases in resampler
+
+2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ docs: Update common. Use upload-doc.mak instead of upload.mak
+
+2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ docs: fix typo
+
+2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: reset accum when dropping samples
+ When we are resampling and we drop samples because we paused, reset the accum
+ counter because it's now invalid.
+
+2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/video/gstbasevideodecoder.h:
+ docs: Fix a couple of warnings from the docs build.
+
+2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/audio/testchannels.c:
+ Don't include config.h multiple times when build audio testchannel app.
+ Fixes build problem on win32 (#585075).
+
+2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2/uridecodebin: Fix connection-speed propagation
+ uridecodebin expects the passed connection-speed value in kbps, so we
+ need to divide the value stored in bps by 1000. Also, lower the upper
+ limit on the properties to the value that we can actually store in our
+ internal guint (which is plenty high enough)
+
+2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ * tests/check/elements/subparse.c:
+ subparse: recognise more subrip timestamp variants
+ Be even less restrictive in what we accept for .srt timestamps when
+ typefinding and parsing subrip subtitles and add a unit test for
+ the 'new' format. Fixes #585197.
+
+2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: add some more docs
+
+2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: Avoid a compiler warning.
+
+2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Updated documentation for GstRTSPResult.
+ Moved GST_RTSP_ELAST to be last in the documentation to match the actual
+ enum values.
+
+2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * autogen.sh:
+ autogen: remove -Wno-portability from here
+ as it is in configure.ac now.
+
+2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Plug a memory leak.
+ Free memory related to any partially read and/or written RTSP messages.
+
+2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: no need to cause discont when clipping
+ Remove the discont-when-clipping hack now that basesink provides us with
+ correctly clipped samples when stepping.
+
+2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: don't align when we clip
+ Don't align samples when they were clipped. Not entirely correct but better than
+ nothing for now.
+
+2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/.gitignore:
+ * tests/examples/seek/stepping2.c:
+ examples: add stepping example in PLAYING
+ Add stepping example in PLAYING, audio is a bit distorted because basesink does
+ not provide good clipping info yet.
+
+2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for hdv/aux-* formats.
+
+2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
+
+ * ext/schroedinger/Makefile.am:
+ Added libgstbase to schro's LIBADD
+ Fixes #585079
+
+2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/tag/gstid3tag.c:
+ libgsttag: don't extract genres from empty ID3v1 tags
+ If we don't have any other info, don't try to interpret the
+ genre field. In particular we don't want to interpret a genre
+ of 0 as 'Blues' if no other fields are set and the entire tag
+ is just empty.
+
+2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: make sure varargs are of right type
+ Explicitly cast the variables to g_object_set to their right types.
+
+2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: increase stream probing queues
+ When we are probing for streams, we want to set the queue size in such a way
+ that we can scan a maximum amount of data without consuming too much memory.
+ Therefore, remove the time limit on the queue and only stop scanning after 2MB
+ of data.
+ See #584104.
+
+2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Fixed a typo.
+
+2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Remove an unused variable.
+
+2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Removed duplicate initialization of conn->writefd.
+
+2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use #defined status codes.
+
+2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Correct gen_tunnel_reply().
+ Prevent gen_tunnel_reply() from generating an incomplete response
+ in case an error response code is given.
+
+2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
+ See #584835. Also update win32 files while we're at it.
+
+2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: API: Add {audio,video,text}-tags-changed signals
+ Fixes bug #584686.
+
+2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: don't put invalid bitrate values into the taglist
+ Bitrates are stored as 32-bit signed integers in the vorbis
+ identification headers, but seem to be read incorrectly,
+ namely as unsigned 32-bit integers, into the vorbis structure
+ members which are of type long, which makes our check for
+ values <= 0 fail with files that put -1 in there for unset
+ values.
+
+2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/.gitignore:
+ ignore: add new stepping app to ignore
+
+2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/stepping.c:
+ examples: add stepping example.
+ Add an example of using playbin2 and frame stepping to simulate variable rate
+ playback based on a sine wave.
+
+2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: also set custom text and subp sinks
+ Set the custom subpicture and text sinks along with the custom audio and video
+ sinks when needed.
+ Fix a little docs blurb too.
+
+2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: add G_LIKELY because we can
+
+2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix caps for ogg typefinder.
+
+2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: remove some cruft from -sections.txt file
+
+2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ * tests/examples/seek/seek.c:
+ add framestepping to playbin2 and seek
+
+2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Avoid compiler warnings with -Wextra.
+
+2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
+
+2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Remove an unused variable.
+
+2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
+
+2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: Have playbin recognise PGS subpicture streams
+ Recognise PGS subpicture streams and connect them to the SPU pad
+ in playsink. Unfortunately this fails badly with negotiation errors
+ if the SPU is not recent enough to support the stream. I'm not sure
+ how to add format negotiation in yet.
+
+2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
+
+2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: fix volume handling for audio sinks without "volume" property
+ When using an audio sink without a "volume" property, volume control
+ would only work for the first song. For the next song, we'd try to
+ re-use the existing audio chain, but inadvertently set chain->volume
+ to NULL instead of to the existing volume element.
+
+2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: cosmetic change to avoid unnecessary line breaks
+ Looks nicer and works around gst-indent silliness.
+
+2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: don't lose the ref to the volume element
+ Only release the ref to the volume element when it is controled by a sink. For
+ software volume we never have to fear that it will change.
+
+2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2: actually use configured audio/video sinks
+ playbin2 inadvertently used autoaudiosink and autovideosink up to now,
+ since it would overwrite the sinks configured via the "audio-sink"
+ and "video-sink" properties with the stream-specific group sinks when
+ configuring the outputs. Those are usually NULL however, so that would
+ overwrite the configured sinks with NULL which makes playbin2 then
+ default to the auto sinks. Fix this by keeping a reference to each
+ configured sink in playbin2 and setting up the right sinks depending
+ on whether there is a stream-specific sink or not.
+ Fixes #584020.
+
+2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/examples/seek/seek.c:
+ seek: add volume label and sync with sink volume
+ Look at the volume and have the pulsemixer open at same time. Unfortunately
+ playbin2 does not emit notify on volume right, so this polls for now.
+
+2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: remove leftover elements
+ Remove all of the elements inside decodebin2 when goint to READY and NULL.
+ Makes decodebin2 reusable.
+ Fixes #583750
+
+2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2; release refs to volume/mute properties
+ Release the refs to the volume and mute property elemens before setting the
+ child elements to READY or NULL.
+ Fixes #583318
+
+2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/gdp/gstgdppay.c:
+ gdppay: set caps on outgoing buffers
+ Set caps on outgoing buffers because NULL caps confuse basetransform.
+ Fixes #583867
+
+2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: also note the order of IP4 addresses
+ IP4 addresses are also stored in network byte order. Make a note of this in the
+ docs.
+
+2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * ext/theora/theoraparse.c:
+ theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
+
+2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
+ This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
+ We now require GLib 2.16.
+
+2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ Update common
+
+2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: document that the port is network order
+ Document the fact that we store the port number in network order in
+ GstNetAddress and that the caller should byteswap appropriately.
+
+2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ videoscale: Add support for 16 bit grayscale in native endianness
+
+2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
+
+2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
- * sys/v4l/gstv4lcolorbalance.h:
- * sys/v4l/gstv4ltuner.h:
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/v4l_calls.h:
- * sys/v4l/videodev_mjpeg.h:
- * tests/check/elements/audioconvert.c:
- * tests/check/elements/audioresample.c:
- * tests/check/elements/audiotestsrc.c:
- * tests/check/elements/videotestsrc.c:
- * tests/check/elements/volume.c:
- * tests/examples/seek/scrubby.c:
+ videotestsrc: Add support for 16 bit grayscale in native endianness
+
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
+
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
+
+2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix boundary case for seeking.
+ When we have exactly 0 bytes left to search, make sure we stop instead of going
+ into an infinite loop.
+
+2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
+
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * gst-libs/gst/cdda/sha1.c:
+ * gst-libs/gst/cdda/sha1.h:
+ cddabasesrc: Remove copy of sha1 digest
+ Remove our copy of sha1 digest now that we depend on glib 2.16.
+ Fixes #536313
+
+2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
+
+ * gst-plugins-base.spec.in:
+ Update spec file
+
+2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ * gst-libs/gst/video/gstbasevideoutils.c:
+ * gst-libs/gst/video/gstbasevideoutils.h:
+ * win32/common/libgstvideo.def:
+ video: don't expose internal gst_adapter_get_buffer() helper function
+ If it's really needed it should go into GstAdapter in core.
+
+2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ basevideo: Fix memleak
+
+2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
+
+ * ext/schroedinger/gstschrodec.c:
+ * ext/schroedinger/gstschroparse.c:
+ schro: Fix usage of adapter_masked_scan_uint32
+ Because *somebody* changed the API without telling me.
+
+2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
+
+ * ext/schroedinger/gstschro.c:
+ schro: Change package name to GST_PACKAGE_NAME
+
+2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ basevideo: Add preset interface to encoder
+
+2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
+
+ * gst/audioresample/gstaudioresample.c:
+ Run liboil benchmark multiple times
+ The statistics function requires multiple runs, otherwise
+ it causes a divide by zero error.
+
+2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * m4/gst-fionread.m4:
+ m4: fix 'suspicious cache value' warning for gst-fionread.m4
+ .. here as well (should really be moved to common, but I'm too lazy).
+
+2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: detect and report errors better
+ Check the return values of a couple more libvorbis functions and post an error
+ when something is wrong instead of continuing and crashing.
+
+2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: fix initial volume and mute handling
+ Use two flags to remember volume/mute changes at times when we don't have the
+ audiochain yet (e.g. construction). Only set values when they were actualy
+ changed. This makes pulseaudio's stream restore functional.
+
+2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
+ Automatic update of common submodule
+ From d3a8fab to 888e0a2
+
+2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * win32/common/libgstvideo.def:
+ win32: Remove gst_adapter_masked_scan_uint32 from the exports
+
+2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: improve debug message
+
+2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
+
+ * gst-libs/gst/tag/gstid3tag.c:
+ gstid3tag: Don't extract a track number unless present.
+ In ID3v1, a track number is present only if byte 125 is null AND
+ byte 126 is non-null. If the track number is not present, don't add
+ a track number tag with value 0.
+
+2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/video/gstbasevideoutils.c:
+ * gst-libs/gst/video/gstbasevideoutils.h:
+ videoutils: remove adapter methods
+ Remove adapter methods now that they are in core.
+
+2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * win32/common/libgstvideo.def:
+ defs: add new symbols
+
+2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ autogen: pass -Wno-portability to automake to suppress warnings
+ GNU make is needed.
+
+2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/.gitignore:
+ gitignore: remove bogus *.sgml wildcard - these files are tracked in git
+
+2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/tcp/gsttcpclientsrc.c:
+ tcpclientsrc: this is not a live source
+ Don't mark us as a live source because we are not.
+
+2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: only send flush_stop when seek failed
+ This is still not the ultimate fix. Added some comment to explain the troubles.
+
+2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: return the return value of wait_preroll
+ Return the value that _wait_preroll() returned instead of always WRONG_STATE.
+
+2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: send flush_stop to match flush_start
+ Adder was relying that something else sends a flush stop. When using adder with
+ a livesource it was not getting a flush_stop and thus all pads downstream where
+ keept flushing. Mark a pending flush_stop and send it when we are working on
+ the new segment back in the streaming thread.
+
+2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
+
* tests/examples/seek/seek.c:
- expand tabs
+ seek: ui improvements
+ Repaint the window black on expose, as this looks nicer when resizing or using
+ the expander. Also show time after slider, as this saves a whole line (nice on
+ small displays).
+
+2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstdecodebin.c:
+ decodebin: use iterators instead of list
+ The list api is deprecated. Use threadsafe iterators instead.
+
+2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: configure caps on decodebin2
+ Implement the caps property by setting the configured caps on new decodebin2
+ objects.
+ Fixes #582749
+
+2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some _caps_ref in some cases
+ Only mess with the caps refcount when we configure different caps.
+
+2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix potential caps leak
+ Free the user-configured caps in finalize.
+
+2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add queue after cdda://
+ Add a queue2 after the raw output pads of certain sources such as those for uris
+ like cdda://
+ No tuning of the queue is done yet as the defaults seem to work fine for me.
+ Fixes #582528
+
+2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't loop when at EOS
+ When we try to read the last page, don't try to read past the upper boundary, as
+ this might cause endless loops.
+ See #582942
+
+2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Don't drain remaining buffers after a flush.
+ If we were resetted (due to a flush), we can not drain the remaining
+ buffers since they would be pushed before a valid new newsegment event.
+
+2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
+
+ * ext/theora/theoradec.c:
+ theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
+
+2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: add more logging and return value checking
+
+2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: handle the return value from iterator_fold
+
+2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: use the pad in logging as objects
+ Helps to differenciate between source and sinks pads.
+
+2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/examples/seek/seek.c:
+ seek: use parser for mp3 and rename variable
+
+2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/seek.c:
+ seek: add playbin2 options in expander
+ Add the playbin2 stream selection options inside an expander to preserve some
+ space on screen.
+
+2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for v210 and v216 formats
+
+2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/gstbasevideocodec.c:
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ video: remove // comments
+
+2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add Y444, v210, v216 formats
+
+2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
+
+ * configure.ac:
+ * ext/Makefile.am:
+ * ext/schroedinger/Makefile.am:
+ * ext/schroedinger/gstschro.c:
+ * ext/schroedinger/gstschrodec.c:
+ * ext/schroedinger/gstschroenc.c:
+ * ext/schroedinger/gstschroparse.c:
+ * ext/schroedinger/gstschroutils.c:
+ * ext/schroedinger/gstschroutils.h:
+ schro: Move schro plugin from Schroedinger
+ Previous history is in Schroedinger. Depends on, and is an example
+ of using, GstBaseVideo* base classes.
+ Code was reindented, and an #ifdef HAVE_ENCODER removed.
+
+2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstbasevideocodec.c:
+ * gst-libs/gst/video/gstbasevideocodec.h:
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideodecoder.h:
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ * gst-libs/gst/video/gstbasevideoencoder.h:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ * gst-libs/gst/video/gstbasevideoparse.h:
+ * gst-libs/gst/video/gstbasevideoutils.c:
+ * gst-libs/gst/video/gstbasevideoutils.h:
+ video: Copy BaseVideo classes from Schroedinger
+
+2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: add num-fds property
+ multifdsink::num-fds
+
+2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
+
+2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Implement Preset interface
+
+2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/theora/theoraenc.c:
+ theoraenc: Implement Preset interface
+
+2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: Implement Preset interface
+
+2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: Fix cdda:// playback
+ Don't send async-start when the playsink has already been configured
+ before changing state.
+
+2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: require core CVS for gst_adapter_prev_timestamp()
+ which is used in the libvisual plugin.
+
+2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * AUTHORS:
+ AUTHORS: fix my email
+
+2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: make our internal time monotonic
+ Make the internal time increase monotonically.
+
+2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/libvisual/visual.c:
+ visual: remove next_ts variable
+ We can remove the next_ts variable as we don't use it anymore.
+
+2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/libvisual/visual.c:
+ visual: use new adapter timestamp code
+ Use the new adapter timestamp tracking code to make things easier and produce
+ vastly better output timestamps.
+
+2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * po/Makevars:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: avoid conflicts of local *.po files with files in git
+ Make it so that filenames and line numbers are only stored in the *.pot file
+ (which is not in git), but not in the individual *.po files. This information
+ is hardly useful for translators in our case, and it should avoid the constant
+ conflicts of local *.po files with the ones in git which are caused by the
+ source files changing and the line numbers being updated. This commit might
+ cause one last merge conflict for you, which you can work around with
+ "git checkout po/*.po" before merging or pulling. After that there should
+ (hopefully) not be any more local modifications of these files (unless
+ someone committed additions or changes to translated strings and the
+ *.po files haven't been updated yet, that is).
+
+2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/audioresample.c:
+ tests: fix audioresample unit test on big endian architectures
+ Don't hardcode endianness=1234 in the filtercaps, it will cause
+ pad link failures which will result in the test timing out.
+
+2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: fix broken enum nick - it should have a hyphen
+ The enum nick should be 'sine-table', not 'sine table'. Technically this is
+ an API/ABI change I guess, but anyone who was using this and didn't report
+ it deserves this.
+
+2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: seek to the requested byte offset, not the expected byte offset
+
+2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: support more than just one channel
+
+2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ propertyprobe: Fix typo in the docs
+
+2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theora.c:
+ * ext/vorbis/vorbis.c:
+ Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
+
+2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ videorate: handle invalid timestamps better
+ Handle buffers with -1 timestamps better by keeping track of the en time of the
+ previous buffer and assuming the -1 timestamp buffer goes right after the
+ previous one.
+ when we have two buffers that are equally good, output the oldest buffer once to
+ minimize latency.
+ don't try to calculate latency when the input framerate is unknown.
+
+2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: small debug statement in DISCONT
-=== release 0.10.0 ===
+2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-12-05 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: fix abuse of ogg API, handle broken oggs
+ When we feed the ogg sync layer, we need to feed it contiguous data even if the
+ sync layer did not consume all of it yet. This makes sure that it always finds
+ the next page even for more corrupted files. Use a different read_offset for
+ this purpose. since we now keep track of the sync layer, we don't have to reset
+ after finding a start of a page.
+ Add some more debug info for the error paths.
+ Only reset the sync layer when we perform a seek operation.
+ Avoid failure when the next chain has no bos pages but instead simply ignore it.
+ when we receive unknown page serial numbers mid stream, don't fail but post a
+ warning and hope that we get back on track later.
+ Fixes #579642
+
+2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.10.0, "Mont-d'or"
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: make subpictures a raw output format
+ Subpictures are a raw format, we want those pads exposed so that playbin2 can do
+ the subpicture mixing.
-2005-12-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/Makefile.am:
- Build fix for when gtk is not available.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ rtpdepay: add some more comments
-2005-12-05 Andy Wingo <wingo@pobox.com>
+2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/libvisual/visual.c: (get_buffer):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer):
- * ext/pango/gsttextrender.c: (gst_text_render_chain):
- * ext/theora/theoradec.c: (theora_handle_data_packet):
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_enc_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- Update for alloc_buffer changes.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: make sure values are ever increasing
-2005-12-05 Andy Wingo <wingo@pobox.com>
+2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- patch by: Kai Vehmanen <kv2004 eca cx>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c
- (gst_base_rtp_depayload_thread): Fix busy loop (#323017).
+ * gst/playback/gstplaysink.c:
+ playbin2: make fallback identity silent
+ Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
+ element so that it consumes less CPU.
-2005-12-04 Andy Wingo <wingo@pobox.com>
+2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- patch by: Sebastien Cote <sebas642 yahoo ca>
-
- * pkgconfig/gstreamer-plugins-base.pc.in (Libs): Add -L flag.
- Fixes #319172.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2: handle custom audiosinks differently
+ Keep track of the autoplugged custom sinks and configure them in the playsink
+ element when we have collected all streams.
+ Also make sure that we only select one custom sink.
+ When unreffing the internal sink, we don't need to change the state to NULL.
-2005-12-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: unify custom sink get/set functions
+ Use one function to set/get all of the different sink types.
+ cleanup up the subpicture chain too.
+ Allow setting a custom subpicture sink.
+
+2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/tunernorm.h:
+ interfaces: Seperate some more struct definitions from typedefs
+
+2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/navigation.h:
+ * gst-libs/gst/interfaces/videoorientation.h:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ interfaces: Seperate some more struct definitions from typedefs
+
+2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * win32/common/libgstinterfaces.def:
+ Add new functions to win32 exports
+
+2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add new functions to the docs
+
+2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixer.h:
+ interfaces: API: Add gst_mixer_get_mixer_type()
+ This is a convenience function that returns the mixer_type
+ of the interface struct.
+
+2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/colorbalance.c:
+ interfaces: Add docs for gst_color_balance_get_balance_type()
+
+2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
+
+ * autogen.sh:
+ Run libtoolize before aclocal
+ This unbreaks the build in some cases. Fixes bug #582021
+
+2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
- * ext/pango/gsttimeoverlay.c:
- Add gtk-doc blurbs to pango elements.
+ textrender: Correctly initialize the background for ARGB too
-2005-12-02 Wim Taymans <wim@fluendo.com>
+2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
- * gst/audioresample/buffer.h:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- * gst/audioresample/resample.c: (resample_input_flush),
- (resample_input_pushthrough), (resample_input_eos),
- (resample_get_output_size_for_input),
- (resample_get_input_size_for_output), (resample_get_output_size),
- (resample_get_output_data):
- * gst/audioresample/resample.h:
- * gst/audioresample/resample_ref.c: (resample_scale_ref):
- Fix audioresample, seek torture, new segments, reverse negotiation
- etc.. work fine.
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ textrender: Use libgstvideo functions to create caps
+ Also check if downstream wants ARGB always when we get
+ new caps.
-2005-12-02 Wim Taymans <wim@fluendo.com>
+2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Small cleanups.
+ * ext/pango/gsttextrender.c:
+ textrender: Don't always use ARGB if downstream supports it but take it's preference
-2005-12-02 Wim Taymans <wim@fluendo.com>
+2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- Post errors.
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ textrender: Add support for ARGB and alignment properties
+ Fixes bug #581571.
-=== release 0.9.7 ===
+2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ * ext/pango/gsttextrender.c:
+ textrender: Add ; after GST_BOILERPLATE to fix indention
- * configure.ac:
- releasing 0.9.7, "Mi Perro No Tiene Ninguna Nariz"
+2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
- * Makefile.am:
- * po/hu.po:
- * win32/MANIFEST:
- * win32/gst.sln:
- add win32 MANIFEST file
- do something to the hungarian translation
+2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
-2005-12-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: made mp3_type_find less aggressive
+ mp3_type_find could suggest already when only a single valid header
+ was found, if it ran out of data before the end of the next frame.
+ Therefore, ignore the last found frame if it was incomplete.
+ Fixes bug #579692.
- * ext/Makefile.am:
- Add $(PANGO_DIR) to SUBDIRS
+2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttimeoverlay.c:
- Fix and improve element descriptions.
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Store cover art in vorbiscomments
+ Fixes bug #513373.
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/inspect/plugin-pango.xml:
- add pango plugin to docs
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/colorbalance.h:
+ interfaces: API: Add gst_color_balance_get_balance_type()
+ This is a convenience function that returns the balance_type
+ of the interface struct.
+
+2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/colorbalancechannel.h:
+ * gst-libs/gst/interfaces/tuner.h:
+ * gst-libs/gst/interfaces/tunerchannel.h:
+ interfaces: Separate struct definitions from typedefs
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ Fix libdir for uninstalled gstreamer-app library
+
+2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for APE tag caps
+
+2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- * ext/Makefile.am:
- moved pango to base
+ configure: bump core requirement to last release
+ as that's more likely to be true than that we need
+ only 0.21.1.
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * common:
* configure.ac:
- * tests/Makefile.am:
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/stress-xoverlay.c: (myclock), (open_display),
- (close_display), (resize_window), (move_window), (create_window),
- (terminate_playback), (pause_playback), (start_playback), (main):
- add stress test for xoverlay from Julien
+ configure: rename CVS -> git in a couple of places
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertppayload.c:
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Do burger's rename for rtp payloaders and depayloaders
+ * configure.ac:
+ configure: bump GLib requirement to GLib >= 2.16
+ as per the New Regime (see wiki).
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/:
- add Visual Studio 6 build files
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: cache events from upstream and re-send them once we have a source pad
+ Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
+ Fixes #580318.
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * gst-libs/gst/interfaces/xoverlay.c:
- * gst-libs/gst/video/gstvideofilter.c:
- * gst-libs/gst/video/gstvideosink.c:
- update documentation
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_base_init),
- (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_init),
- (gst_multi_fd_sink_finalize), (gst_multi_fd_sink_add),
- (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
- (gst_multi_fd_sink_get_stats),
- (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_handle_client_read),
- (gst_multi_fd_sink_client_queue_data),
- (gst_multi_fd_sink_client_queue_caps),
- (gst_multi_fd_sink_client_queue_buffer),
- (gst_multi_fd_sink_new_client),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_recover_client),
- (gst_multi_fd_sink_queue_buffer),
- (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_thread),
- (gst_multi_fd_sink_render), (gst_multi_fd_sink_set_property),
- (gst_multi_fd_sink_get_property), (gst_multi_fd_sink_start),
- (gst_multi_fd_sink_stop), (gst_multi_fd_sink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_get_type),
- (gst_tcp_client_sink_base_init), (gst_tcp_client_sink_class_init),
- (gst_tcp_client_sink_init), (gst_tcp_client_sink_finalize),
- (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
- (gst_tcp_client_sink_set_property),
- (gst_tcp_client_sink_get_property), (gst_tcp_client_sink_start),
- (gst_tcp_client_sink_stop), (gst_tcp_client_sink_change_state):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_base_init),
- (gst_tcp_client_src_class_init), (gst_tcp_client_src_init),
- (gst_tcp_client_src_finalize), (gst_tcp_client_src_getcaps),
- (gst_tcp_client_src_create), (gst_tcp_client_src_set_property),
- (gst_tcp_client_src_get_property), (gst_tcp_client_src_start),
- (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpplugin.c: (plugin_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_base_init),
- (gst_tcp_server_sink_class_init), (gst_tcp_server_sink_init),
- (gst_tcp_server_sink_finalize),
- (gst_tcp_server_sink_handle_server_read),
- (gst_tcp_server_sink_removed), (gst_tcp_server_sink_handle_wait),
- (gst_tcp_server_sink_set_property),
- (gst_tcp_server_sink_get_property),
- (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_base_init),
- (gst_tcp_server_src_class_init), (gst_tcp_server_src_init),
- (gst_tcp_server_src_finalize), (gst_tcp_server_src_create),
- (gst_tcp_server_src_set_property),
- (gst_tcp_server_src_get_property), (gst_tcp_server_src_start),
- (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
- * gst/tcp/gsttcpserversrc.h:
- more borgifying
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: support UYVY raw 4:2:2 in riff.
- * docs/plugins/Makefile.am:
+2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ Back to development -> 0.10.23.1
+
+2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
+
+ * ext/theora/theoradec.c:
+ theoradec: fix buffer overrun on 422 decode.
+
+2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
+
+ * ext/theora/theoradec.c:
+ theoradec: 444 support.
+
+2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
+
+ * ext/theora/theoradec.c:
+ theoradec: handle 422 images (as YUY2).
+
+2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
+
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theoradec: rearrange code in preparation for 422 and 444 support.
+
+=== release 0.10.23 ===
+
+2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
- * gst/audioconvert/plugin.h:
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_get_type),
- (gst_audio_rate_base_init), (gst_audio_rate_class_init),
- (gst_audio_rate_setcaps), (gst_audio_rate_init),
- (gst_audio_rate_chain), (gst_audio_rate_set_property),
- (gst_audio_rate_get_property), (gst_audio_rate_change_state),
- (plugin_init):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_base_init),
- (gst_audio_test_src_class_init), (gst_audio_test_src_init),
- (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
- (gst_audio_test_src_get_query_types),
- (gst_audio_test_src_src_query), (gst_audio_test_src_create_sine),
- (gst_audio_test_src_create_square),
- (gst_audio_test_src_create_saw),
- (gst_audio_test_src_create_triangle),
- (gst_audio_test_src_create_silence),
- (gst_audio_test_src_create_white_noise),
- (gst_audio_test_src_init_pink_noise),
- (gst_audio_test_src_generate_pink_noise_value),
- (gst_audio_test_src_create_pink_noise),
- (gst_audio_test_src_change_wave), (gst_audio_test_src_get_times),
- (gst_audio_test_src_create), (gst_audio_test_src_set_property),
- (gst_audio_test_src_get_property), (gst_audio_test_src_start),
- (plugin_init):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- * gst/subparse/gstsubparse.c: (gst_sub_parse_get_type),
- (gst_sub_parse_base_init), (gst_sub_parse_class_init),
- (gst_sub_parse_init), (gst_sub_parse_formats),
- (gst_sub_parse_src_eventmask), (gst_sub_parse_src_event),
- (convert_encoding), (get_next_line),
- (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (feed_textbuf), (handle_buffer),
- (gst_sub_parse_loop), (gst_sub_parse_chain),
- (gst_sub_parse_change_state), (gst_sub_parse_type_find),
- (plugin_init):
- * gst/subparse/gstsubparse.h:
- * gst/videorate/gstvideorate.c: (gst_video_rate_get_type),
- (gst_video_rate_base_init), (gst_video_rate_class_init),
- (gst_video_rate_transformcaps), (gst_video_rate_getcaps),
- (gst_video_rate_setcaps), (gst_video_rate_blank_data),
- (gst_video_rate_init), (gst_video_rate_event),
- (gst_video_rate_chain), (gst_video_rate_set_property),
- (gst_video_rate_get_property), (gst_video_rate_change_state),
- (plugin_init):
- * gst/videoscale/gstvideoscale.c:
- (gst_video_scale_method_get_type), (gst_video_scale_get_capslist),
- (gst_video_scale_src_template_factory),
- (gst_video_scale_sink_template_factory),
- (gst_video_scale_get_type), (gst_video_scale_base_init),
- (gst_video_scale_class_init), (gst_video_scale_init),
- (gst_video_scale_set_property), (gst_video_scale_get_property),
- (gst_video_scale_transform_caps), (gst_video_scale_get_format),
- (gst_video_scale_prepare_size), (parse_caps),
- (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
- (gst_video_scale_fixate_caps), (gst_video_scale_prepare_image),
- (gst_video_scale_transform), (gst_video_scale_handle_src_event),
- (plugin_init):
- * gst/videoscale/gstvideoscale.h:
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_video_test_src_pattern_get_type),
- (gst_video_test_src_base_init), (gst_video_test_src_class_init),
- (gst_video_test_src_init), (gst_video_test_src_src_fixate),
- (gst_video_test_src_set_pattern),
- (gst_video_test_src_set_property),
- (gst_video_test_src_get_property), (gst_video_test_src_getcaps),
- (gst_video_test_src_parse_caps), (gst_video_test_src_setcaps),
- (gst_video_test_src_event), (gst_video_test_src_get_times),
- (gst_video_test_src_create), (plugin_init):
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_get_size),
- (gst_video_test_src_smpte), (gst_video_test_src_snow),
- (gst_video_test_src_black):
- * gst/videotestsrc/videotestsrc.h:
- borgify further
- clean up docs a little
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.23
+
+2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+
+2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.22.6 pre-release
+
+2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: fix resume after pause
+ Don't ignore the state change of the children, they might be doing an ASYNC
+ state change.
+
+2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ChangeLog:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.22.5 pre-release
-2005-11-30 Wim Taymans <wim@fluendo.com>
+2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_event), (gst_basertppayload_push),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- closed #320644.
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcp-marshal.list:
+ multifdsink: fix signature of the add-full signal
+ The second parameter is a GstSyncMethod enum, not a boolean.
-2005-11-30 Julien MOUTTE <julien@moutte.net>
+2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/video/gstvideofilter.c:
- * gst-libs/gst/video/gstvideosink.c:
- * gst-libs/gst/video/gstvideosink.h: Adding docs.
+ * gst/playback/gstplaysink.c:
+ playsink: initialize variable too
+
+2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ * gst/playback/gstplaysink.c:
+ playbin2: make playsink go ASYNC to PAUSED
+ Make playsink go async to the PAUSED state instead of relying on uridecodebin
+ for async behaviour in playbin. This solves some problems (mainly with DVD)
+ where the pipeline would go to PLAYING before preroll completed, failing to
+ select the audiosink clock.
+ Fixes #581727
+
+2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
- * LICENSE:
- move
+ * configure.ac:
* po/af.po:
* po/az.po:
+ * po/bg.po:
+ * po/ca.po:
* po/cs.po:
+ * po/da.po:
+ * po/de.po:
* po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
* po/hu.po:
+ * po/id.po:
* po/it.po:
+ * po/ja.po:
+ * po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
- * Makefile.am:
- update
- * scripts/autoplugins.sh:
- remove
-
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * Makefile.am:
- * configure.ac:
- * examples/Makefile.am:
- * examples/capsfilter/Makefile.am:
- * examples/capsfilter/capsfilter1.c:
- * examples/gob/Makefile.am:
- * examples/gob/gst-identity2.gob:
- * examples/indexing/.cvsignore:
- * examples/indexing/Makefile.am:
- * examples/indexing/indexmpeg.c:
- * examples/seeking/.cvsignore:
- * examples/seeking/Makefile.am:
- * examples/seeking/cdparanoia.c:
- * examples/seeking/cdplayer.c:
- * examples/seeking/chained.c:
- * examples/seeking/scrubby.c:
- * examples/seeking/seek.c:
- * examples/stats/Makefile.am:
- * examples/stats/mp2ogg.c:
- * examples/switch/.cvsignore:
- * examples/switch/Makefile.am:
- * examples/switch/switcher.c:
- * tests/Makefile.am:
- * tests/check/generic/.cvsignore:
- * tests/check/pipelines/.cvsignore:
- * tests/examples/Makefile.am:
- * tests/examples/seek/Makefile.am:
- reorganize stuff under tests/
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- Go away you stupid GstStaticPadTemplate memleak.
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/net/README:
- * gst-libs/gst/net/gstnetbuffer.c:
- * gst-libs/gst/net/gstnetbuffer.h:
- this was moved to "netbuffer"
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_get_type),
- (gst_video_filter_class_init), (gst_video_filter_init):
- * gst-libs/gst/video/gstvideofilter.h:
- borgify name to bring in line with other classes
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioscale/.cvsignore:
- * gst/audioscale/Makefile.am:
- * gst/audioscale/README:
- * gst/audioscale/audioscale.vcproj:
- * gst/audioscale/dtof.c:
- * gst/audioscale/dtos.c:
- * gst/audioscale/functable.c:
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/gstaudioscale.h:
- * gst/audioscale/private.h:
- * gst/audioscale/resample.c:
- * gst/audioscale/resample.h:
- * gst/audioscale/test.c:
- remove
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst-libs/gst/netbuffer/Makefile.am:
- really, really tired
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Update for new GstTypeFindFactory _register()
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
- Let's not override libgstnet from core for no reason...
- (libgstnetbuffer_@GST_MAJORMINOR@_la_SOURCES):
- Ok, maybe not so quick next time.
-
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- moved gst-libs/gst/net to netbuffer through CVS surgery
- remove old directory
- updating build to accomodate
- (#322257)
-
-2005-11-29 Andy Wingo <wingo@pobox.com>
-
- * pkgconfig/gstreamer-plugins-base.pc.in:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
- (#322257).
-
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gststreamselector.c: (gst_stream_selector_chain):
- 3rd time's the charm. Correct ref-counting for discarded buffers.
-
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init),
- (gst_stream_selector_set_property),
- (gst_stream_selector_get_property),
- (gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
- Fix ref-counting
-
-2005-11-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/subparse/gstsubparse.c: (feed_textbuf):
- Don't access already unref'ed buffer.
-
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init), (gst_stream_selector_init),
- (gst_stream_selector_dispose), (gst_stream_selector_set_property),
- (gst_stream_selector_get_property),
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
- * gst/playback/gststreamselector.h:
- Add the active-pad property for playbin to use shortly. Ignore buffers
- from any other pad, returning GST_FLOW_NOT_LINKED
-
-2005-11-29 Julien MOUTTE <julien@moutte.net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find): Free the list,
- patch from bug #322704 (Alessandro Decina).
-
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/Makefile.am:
- folded audiofilter into the audio library
-
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/gstvideoscale.c:
- remove unimplemented scale methods
-
-2005-11-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
- Don't leak caps.
-
-2005-11-28 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
- (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_setcaps): Fixed a tricky bug. When caps renegotiation
- happens (only visible on ximagesink but bug is in xv too) set_caps was
- destroying the internal x[v]image used to memcpy non locally alloced
- buffers so that it got renewed on next _chain. The issue is that
- _expose will try to put that image as it reffed it in _put.
- Using gst_buffer_unref instead of destroy fixes it !
-
-2005-11-28 Edward Hervey <edward@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (try_to_link_1), (queue_filled_cb):
- Better use of the queues. Start with a small size queue and only increase
- the size of the queues when the other queues are empty.
-
-2005-11-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/video/Makefile.am:
- compile in copied-over videofilter into the video library
- * gst-libs/gst/video/videosink.h:
- rename the header to gstvideosink.h since it's a base GstObject class
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h:
- use the new header
-
-2005-11-28 Wim Taymans <wim@fluendo.com>
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.22.4 pre-release
- * gst/playback/gstplaybasebin.c: (group_commit), (probe_triggered):
- * gst/playback/gstplaybasebin.h:
- Prepare to handle errors betters.
+2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
- * gst/playback/gstplaybin.c: (add_sink), (setup_sinks):
- Set sinks to PAUSED first before adding and linking them so that
- we don't interrupt dataflow.
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
+ With vorbisenc, compute the granulepos with running time and clip incoming
+ buffers to segment.
+ With theoraenc, drop out of segment buffers.
-2005-11-28 Wim Taymans <wim@fluendo.com>
+2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/TODO:
- Updated TODO
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer size transformations
+ When calculating the input/output buffer sizes in the transform_size function,
+ take the number of channels into account, so we don't end up calculating
+ a buffer size that only contains a partial number of audio frames.
+ Also, when going from output size to input size, round down rather than
+ up, so as to calculate the minimum number of samples that *might* yield
+ a buffer of the intended destination size.
+ Fixes: #580470 and #580952
+
+2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/vorbis/gstvorbisenc.h:
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Ensure output buffers fall within the segment
+ Add the start position of the first segment to the running time
+ used to generate buffer timestamps in vorbisenc. This avoids generating
+ buffers which fall outside the initial segment. The element segment
+ handling requires more extensive fixing, but this at least prevents
+ regressions. Fixes: #580020
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_open_device),
- (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release):
- Small cleanups.
+2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
* gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- Slave to the master clock when going to PLAYING and unslave when
- going to PAUSED.
-
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
- (gst_ring_buffer_acquire), (gst_ring_buffer_release),
- (gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
- (gst_ring_buffer_clear_all), (wait_segment),
- (gst_ring_buffer_commit), (gst_ring_buffer_read),
- (gst_ring_buffer_advance):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add some docs and cleanups.
-
-2005-11-28 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Fix navigation events
- coordinates translation with pixel aspect ratios.
-
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+ Revert "add can-activate-pull property to baseaudiosink"
+ This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
- Use calculated video geometry from _setcaps instead of buffer
- caps to respect pixel aspect ratio. (fixes #322388)
+2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
-2005-11-28 Julien MOUTTE <julien@moutte.net>
-
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstxoverlay.sgml:
- * gst-libs/gst/interfaces/xoverlay.c: Add docs for the Xoverlay
- interface.
-
-2005-11-28 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Refuse to create an XvImage if we can't find the format.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "[baseaudiosink] add docs for can-activate-pull"
+ This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
-2005-11-28 Edward Hervey <edward@fluendo.com>
+2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_template_caps):
- Add ATRAC3 to the list of riff-possible audio caps.
- I know we still don't have a plugin for atrac3, but it's saner to output
- that than a cryptic mimetype.
+ [baseaudiosink] add docs for can-activate-pull
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
+ can-activate-pull.
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds):
- Don't try to create a zero-sized subbuffer.
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
-2005-11-27 Julien MOUTTE <julien@moutte.net>
+2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_free),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
- (gst_ximagesink_bufferpool_clear), (gst_ximagesink_buffer_alloc),
- (gst_ximagesink_expose): Fixed a tricky race.
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_expose): Fixed a tricky race.
- * sys/xvimage/xvimagesink.h:
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ videorate: clear discont on duplicated buffers
+ When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
+ the first pushed buffer but fails to clear it for subsequent buffers. This
+ causes theoraenc!oggmux and possibly other elements to consider this a discont
+ stream.
+ Fix videorate to produce discont as the first buffer and after a flushing seek.
+ Fixes #580271.
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (close_pad_link), (try_to_link_1):
- Remove unused properties, and add queues between demuxers and decoders
- so that a lot more files can preroll properly.
+ * tests/check/Makefile.am:
+ check: Disable the playbin2 for this release, as it is a bit racy.
+ Disable the test, as per the discussion in #580120. Needs re-enabling
+ after the release, when playbin2 is fixed.
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- remove silly include
- * gst/tags/Makefile.am:
- * gst/tags/gsttagediting.c:
- * gst/tags/gsttageditingprivate.h:
- * gst/tags/tagedit.vcproj:
- remove directory, is as good as empty
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
+ The 2s limit is way too small for a lot of files (which have an interleave
+ in time of between 3 and 5s). Instead, leave it to the initial 5s value
+ and reduce the other limits (allowing us to stay memory-efficient).
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
- added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
- * gst-libs/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- and use them
-
-2005-11-27 Julien MOUTTE <julien@moutte.net>
-
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstxoverlay.sgml:
- * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h: More and more documentation.
-
-2005-11-26 Julien MOUTTE <julien@moutte.net>
-
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstxoverlay.sgml: Add the XOverlay interface
- to documentation.
-
-2005-11-26 Julien MOUTTE <julien@moutte.net>
-
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * sys/xvimage/xvimagesink.c: Added doc for xvimagesink.
-
-2005-11-26 Julien MOUTTE <julien@moutte.net>
-
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-playbin.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml: Updated version.
- * sys/ximage/ximagesink.c: Adding documentation for ximagesink.
-
-2005-11-26 Edward Hervey <edward@fluendo.com>
-
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_chain):
- Properly return GstFlowReturn from gst_pad_push in chain functions.
-
-2005-11-25 Michael Smith <msmith@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_convert),
- (gst_ogg_demux_chain_peer), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
- (gst_ogg_demux_send_event), (gst_ogg_demux_loop):
- Handle various conditions better when we don't understand a stream.
- Removes a heap of CRITICALs on ogg streams containing unknown data.
-
-2005-11-24 Andy Wingo <wingo@pobox.com>
-
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_handle_client_write):
- Be threadsafe.
-
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.22.3 pre-release
- * configure.ac: back to HEAD
+2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
-=== release 0.9.6 ===
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix unused variable in compilation with --disable-gst-debug
+ Fixes: #579668
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
- * configure.ac:
- releasing 0.9.6, "White Eight"
+ * common:
+ Automatic update of common submodule
+ From b3941ea to 6ab11d1
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- * docs/plugins/inspect/plugin-sine.xml:
- remove sinesrc some more
+ * gst/playback/gstplaybasebin.c:
+ playbin: only use raw_decoding_mode when it's true
+ First check the pad caps if they are raw before setting the raw_decoding_mode to
+ TRUE. Fixes playback of transport streams and other streams that require large
+ queues.
+ Fixes #579734
-2005-11-23 Wim Taymans <wim@fluendo.com>
+2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
- If we are reading too slowly, jump forward in the ringbuffer
- instead of blocking.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
+ Don't use REPLACE_ALL merge mode when that's not really what we want,
+ as now that REPLACE_ALL actually does what it's supposed to do in
+ core, we drop tags we wanted to keep, such as the various disc id
+ tags. Add unit test for this as well. Fixes #579463.
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_src_setcaps), (get_buffer),
- (gst_visual_chain):
- * ext/theora/theoraenc.c: (theora_enc_sink_setcaps):
- * gst/videorate/gstvideorate.c: (gst_videorate_setcaps),
- (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate), (gst_videotestsrc_create):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get_fps),
- (gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query),
- (gst_v4lmjpegsrc_get), (gst_v4lmjpegsrc_getcaps):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get_fps_list), (gst_v4lsrc_buffer_new):
- Updates for API changes
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.16 API, we require only 2.14
+ Fixes #579267.
-2005-11-23 Wim Taymans <wim@fluendo.com>
+2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Fix for calibration API change.
-
-2005-11-23 Michael Smith <msmith@fluendo.com>
+ baseaudiosink: don't unparent the ringbuffer
+ when going to NULL, don't unparent the ringbuffer because we don't support going
+ back to 0 very well yet.
+ Fixes #579203
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_get_channel_positions),
- (gst_audio_set_channel_positions),
- (gst_audio_set_structure_channel_positions_list),
- (gst_audio_fixate_channel_positions):
- Use gst_value_array_*() functions on value arrays, not
- gst_value_list_*().
-
-2005-11-23 Edward Hervey <edward@fluendo.com>
+2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
- * autogen.sh:
- eheh, sinesrc is gone. Replacing sample srcfile by existing one.
- Fixes autogen
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ RTCP: don't fail when retrieving invalid PT
+ We can't meaningfully assert on valid packet types so just return the type as it
+ is. Update the comments to reflect this.
+ Fixes #579192.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/Makefile.am:
- * check/elements/videotestsrc.c: (setup_videotestsrc),
- (cleanup_videotestsrc), (GST_START_TEST), (videotestsrc_suite),
- (main):
- add a test for videotestsrc
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.h:
+ * gst-libs/gst/app/gstappsrc.h:
+ app: add trivial cast macros
+ Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
+ and add the macros to the standard macros in the docs.
+ Fixes #579130
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/.cvsignore:
- * gst/sine/Makefile.am:
- * gst/sine/gstsinesrc.c:
- * gst/sine/gstsinesrc.h:
- * gst/sine/sinesrc.vcproj:
- and remove sinesrc from the repository. Closes #321446
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ pkgconfig: add the app/ directory to Libs
+ Add the appsrc/appsink directory to the Libs in the uninstalled
+ pkgconfig file so that one can build against it.
+ Fixes #579129
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
- * gst-plugins-base.spec.in:
- remove sinesrc from the build
+ 0.10.22.2 pre-release
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * check/Makefile.am:
- * check/elements/audiotestsrc.c: (setup_audiotestsrc),
- (cleanup_audiotestsrc), (GST_START_TEST), (audiotestsrc_suite),
- (main):
- add a test for audiotestsrc, testing all waves. Even seems
- leak-free at first glance, nice job Stefan
+ * ChangeLog:
+ ChangeLog: regenerate changelog with the gen-changelog script
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
* po/af.po:
* po/az.po:
+ * po/bg.po:
+ * po/ca.po:
* po/cs.po:
+ * po/da.po:
+ * po/de.po:
* po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
* po/hu.po:
+ * po/id.po:
* po/it.po:
+ * po/ja.po:
+ * po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
- Translation string updates
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate), (gst_v4lsrc_get_caps),
- (gst_v4lsrc_set_caps):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get_fps_list), (gst_v4lsrc_buffer_new):
- * sys/v4l/v4lsrc_calls.h:
- Improve v4lsrc, by making it work again.
-
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/libvisual/visual.c: (gst_visual_chain):
- Fix the fps calculations.
+ * po/zh_CN.po:
+ po: Update po files from TP
- * gst/ffmpegcolorspace/avcodec.h:
- Move structure element for clarity
+2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_init):
- * gst-libs/gst/interfaces/tunernorm.h:
- * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_src_link):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_base_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_base_init),
- (gst_v4lmjpegsrc_get_fps), (gst_v4lmjpegsrc_src_convert),
- (gst_v4lmjpegsrc_src_query), (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_getcaps):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate), (gst_v4lsrc_get_caps),
- (gst_v4lsrc_set_caps):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get_fps_list), (gst_v4lsrc_buffer_new):
- * sys/v4l/v4lsrc_calls.h:
- Fractional framerates...
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ win32: Update win32 build files
-2005-11-22 Wim Taymans <wim@fluendo.com>
+2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- And we provide a clock by default, of course...
+ * tests/check/libs/video.c:
+ check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
-2005-11-22 Wim Taymans <wim@fluendo.com>
+2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
- This clock can be slaved to a master clock now.
+ * tests/check/elements/playbin2.c:
+ check: Fix the input uri in playbin2 test.
+ Don't try and use a random file in wim's home directory as a test input
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
- (gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
- (gst_base_audio_sink_set_clock),
- (gst_base_audio_sink_set_property),
- (gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- Handle slaving the internal clock to the clock selected in the
- pipeline.
- Add property to make the basesink not provide a clock.
+2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_wait):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- We can use the clock in GstElement, no need to store it ourselves.
+ * gst-libs/gst/video/video.h:
+ video: Fix typo in the docs
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/tmpl/gstaudio.sgml:
- update
- * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
- (gst_paranoia_endian_get_type):
- * ext/theora/theoraenc.c: (gst_border_mode_get_type):
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
- * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_sync_method_get_type), (gst_unit_type_get_type),
- (gst_client_status_get_type), (gst_multifdsink_class_init),
- (gst_multifdsink_new_client), (gst_multifdsink_recover_client),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_method_get_type):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_pattern_get_type):
- remove deprecated properties
- fix up enums to correctly have short lowercase dashed nicks
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for YVYU YUV colorspace
-2005-11-22 Michael Smith <msmith@fluendo.com>
+2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- Add underscore.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * gst-libs/gst/fft/gstfft.c:
+ docs: fix hyperlink and move fft attribution to the right place
-2005-11-22 Michael Smith <msmith@fluendo.com>
+2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- Use utility method for scaling clocktime for fractional framerates.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ log: use G_GUINT64_FORMAT instead of llu
-2005-11-22 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
- * ext/libvisual/visual.c: (gst_visual_src_setcaps), (get_buffer),
- (gst_visual_chain):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- * ext/theora/theoradec.c: (theora_handle_type_packet):
- * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
- (theora_enc_chain):
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- * gst-libs/gst/video/video.c: (gst_video_frame_rate):
- * gst-libs/gst/video/video.h:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_set_caps):
- * gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_setcaps), (gst_videorate_blank_data),
- (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate), (gst_videotestsrc_getcaps),
- (gst_videotestsrc_parse_caps), (gst_videotestsrc_setcaps),
- (gst_videotestsrc_event), (gst_videotestsrc_create):
- * gst/videotestsrc/gstvideotestsrc.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_setcaps), (gst_ximagesink_change_state),
- (gst_ximagesink_get_times), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
- (gst_xvimagesink_get_times), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h:
- Convert elements to use fractions for their framerate.
- V4L elements to come later tonight.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ RTSP: add missing headers for WMS RTSP
+ Add missing headers related to Windows Media RTSP extension.
+ Fixes #578942
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- remove some deprecated functions
+ * docs/design/draft-keyframe-force.txt:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theoraenc: implement upstream keyframe force
+ Implement handling of upstream keyframe forcing.
+ Update the design documents too.
+ Fixes #578656
-2005-11-22 Andy Wingo <wingo@pobox.com>
+2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
- * Update for gst_tag_setter API changes.
+ * ext/theora/theoraenc.c:
+ theoraenc: factor out keyframe forcing
+ See #578656
-2005-11-22 Andy Wingo <wingo@pobox.com>
+2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggdemux.c (gst_ogg_pad_submit_packet)
- (gst_ogg_demux_perform_seek):
- * ext/theora/theoradec.c (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c (vorbis_dec_sink_event): Run
- update-funcnames.
+ * AUTHORS:
+ * gst-libs/gst/fft/gstfft.c:
+ Give credit to Mark Borgerding (kissfft author)
+ and add myself to AUTHORS as well. Fixes #575638.
-2005-11-22 Wim Taymans <wim@fluendo.com>
+2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
- * examples/seeking/seek.c: (main):
- Give higher priority to bus signals than the gtk events
- to fix a race condition in the segment looping.
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ multifdsink: add property to resend streamheaders
+ Adds a new property in multifdsink, resend-streamheader.
+ If this property is false, the multifdsink will not send the streamheader if
+ there's already one set for a particular client.
+ There are some formats in which every stream needs to start with a certain
+ blob, but you can't inject this blob at leisure. If the producer wants to
+ change the blob in question and sets in as the streamheader on the outgoing
+ buffers' caps, new clients of multifdsink will get the new streamheader, but
+ old clients will break, because they'll see the blob in the middle of the
+ stream.
+ The property is true by default, so existing code will not see any difference.
+ Fixes #578118.
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-plugins-base.spec.in:
- Rename libgsttagedit to libgsttag (#322117).
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ multifdsink: add property to handle client write
+ Add a property to disable listening to client writes. This property is usefull
+ when other code will deal with reading from the client socket.
+ API: GstMultiFdSink::handle-read property
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- Call gst_x_overlay_prepare_xwindow_id() to give applications
- a final chance to set their own xwindow id before the video
- sink creates its own window.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ * win32/common/libgstrtp.def:
+ RTCP: add beginnings of Feedback messages
+ Add the beginnings of parsing and constructing Feedback messages.
+ Fixes #577610.
-2005-11-22 Julien MOUTTE <julien@moutte.net>
+2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Handle navigation
- events correcly with borders if applicable.
+ * gst/playback/gstplaysink.c:
+ playbin2: clear the target
+ Clear the target of our ghostpads before we remove the pad from the element.
+ This to make sure that the internal pad is not left linked to whatever pad we
+ were ghosted to. This should only be a problem when we leak the ghostpads.
+ Also release our subpicture pads.
+ Fixes #577288.
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
- Patch by: Luca Ognibene
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
+ Fixes #570768.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info):
- * gst/ffmpegcolorspace/imgconvert.c:
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Forward-port fixes from the 0.8 branch (patch by Luca Ognibene,
- #318353); use gst_structure_has_name().
-
-2005-11-22 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
- (gst_ximagesink_class_init): Add debug macros on functions.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xwindow_draw_borders),
- (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xwindow_update_geometry),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_event_thread), (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_change_state), (gst_xvimagesink_show_frame),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose), (gst_xvimagesink_set_property),
- (gst_xvimagesink_get_property), (gst_xvimagesink_finalize),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h: Refactoring to handle aspect ratio,
- expose while being PAUSED, out of data flow navigation events, etc..
-
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
- * gst-libs/gst/audio/audio.h:
- fix prototype - wondering why the test worked regardless
+2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: adjust the internal timestamp
+ Adjust the internal timestamp before comparing it against the adjusted clock
+ time.
+ Fixes #578506
- * check/Makefile.am:
- * gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
- * gst-libs/gst/audio/audio.h:
- add a method that returns a proper GstClockTime
+2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: use new clock time methods
+ Use the unadjusted internal clock times to calculate the internal/external
+ offset when calibrating the clock.
+ When going to NULL, unparent and free the ringbuffer, like we do in the source
+ element.
+ See #578506
- * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_base_init),
- (gst_x_overlay_got_xwindow_id), (gst_x_overlay_prepare_xwindow_id):
- * gst-libs/gst/interfaces/xoverlay.h:
- Remove everything having to do with the desired size; add
- gst_x_overlay_prepare_xwindow_id() function; remove the
- 'have-xwindow-id' signal and make gst_x_overlay_got_xwindow_id()
- post a message on the bus instead (#321816).
+2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
- (gst_xvimagesink_xoverlay_init):
- Remove desired size stuff (#321816).
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * win32/common/libgstaudio.def:
+ audioclock: add methods for the internal offset
+ Add two methods for getting the unadjusted time of the clock and one for
+ adjusting an internal time. We will need these methods for correctly handling
+ the time after a gst_audio_clock_reset().
+ Add a debug category and some debug lines to the audio clock.
+ API: gst_audio_clock_get_time()
+ API: gst_audio_clock_adjust()
+ API: GST_AUDIO_CLOCK_CAST()
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (mpeg2_sys_type_find), (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find):
- Terminate vararg functions with NULL instead of 0 to
- make gcc4 happy.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix up the debugs and warnings
+ Use _OBJECT variants because we can. Go over some log statements and put them in
+ the right category.
+ Fixes #567740.
-2005-11-21 Andy Wingo <wingo@pobox.com>
+2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
- patch by: Sebastien Cote <sebas642@yahoo.ca>
-
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- * gst-libs/gst/rtp/gstrtpbuffer.c
- (gst_rtpbuffer_get_payload_buffer): New function. Fixes #321451.
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: fix error in sync-method
+ Multifdsink did not handle sync-method=latest-keyframe correctly when the
+ soft-limit is set to -1 (unlimited).
+ Fixes #578583.
-2005-11-21 Andy Wingo <wingo@pobox.com>
+2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c (gen_audio_element)
- (gen_video_element): Use the new MISSING_PLUGIN core error
- category. Closes #320060.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: use the internal clock time
+ We can't assume that the internal clock time is the same as the function we
+ installed on our provided clock because somebody might have changed it.
- * ext/vorbis/vorbisdec.c (vorbis_dec_sink_event):
- * gst/videorate/gstvideorate.c (gst_videorate_event):
- * ext/theora/theoradec.c (theora_dec_sink_event):
- * ext/theora/theoraenc.c (theora_enc_sink_event): Don't take
- stream lock.
+2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/gstsubparse.c (gst_subparse_src_event):
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek): Update for
- stream lock changes.
+ * tests/examples/seek/seek.c:
+ seek: handle clock-lost messages
+ When we receive a clock-lost message we need to pause and play to select a new
+ clock.
-2005-11-21 Wim Taymans <wim@fluendo.com>
+2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
- (gst_base_audio_sink_provide_clock),
- (gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- * gst/audioresample/gstaudioresample.c:
- Segment update fix.
+ * tests/check/Makefile.am:
+ * tests/check/elements/playbin2.c:
+ check: add a unit test for playbin2
+ Add unit test for playbin2 and include the refcount test in #577794.
-2005-11-21 Andy Wingo <wingo@pobox.com>
+2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * *.h:
- * *.c: Ran scripts/update-macros. Oh yes.
+ * gst/playback/gstplaysink.c:
+ playbin2: fix refcounting of visualisations
+ See #577794.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/Makefile.am:
- * sys/ximage/ximage.c:
- Rename ximage plugin to ximagesink (#321426) (Don't forget to
- remove your old libgstximage.* manually if necessary).
+ * gst/playback/gstplaysink.c:
+ playsink: fix refcounting of custom elements
+ Sink the custom sinks, let other elements we create be sunken by the bin we add
+ them to.
+ Fixes #577794.
-2005-11-21 Michael Smith <msmith@fluendo.com>
+2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_setup):
- Minimal fix for bug #320200: set the min/max bitrate in the correct
- units. A better fix would be to upgrade to the RATEMANAGE2
- interface, rather than using the deprecated interface used here, but
- that would require an update in our libvorbis dependency (to 1.1),
- which is probably undesirable.
+ * tests/check/elements/appsink.c:
+ check: fix appsink test
+ Fix the appsink test now that the method signature changed.
-2005-11-21 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/libvisual/visual.c: (get_buffer):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_fixate):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_caps):
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiotestsrc_src_fixate):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
- * gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
- Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
- (#322027)
+ * gst/playback/gstplaybin2.c:
+ playbin2: handle missing input-selector
+ Gracefully degrade and disable stream selection when input-selector is
+ missing.
+2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ appsink: make callbacks return GstFlowReturn
+ Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
+ errors can be reported properly.
+ Fixes #577827.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_iavs_caps):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_file_header),
- (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
- (gst_riff_parse_info):
- * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_caps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- Fixes for GST_FOURCC_FORMAT API change.
+2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-21 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ ringbuffer: allow for custom commit functions
+ Allow subclasses to override the commit method.
- patch by: Alessandro Dessina <alessandro nnva org>
+2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_set_header_on_caps):
- * ext/ogg/gstoggparse.c (gst_ogg_parse_append_header)
- (gst_ogg_parse_chain):
- * ext/theora/theoraenc.c (theora_set_header_on_caps):
- * ext/vorbis/vorbisenc.c (gst_vorbisenc_set_header_on_caps):
- * ext/vorbis/vorbisparse.c (vorbis_parse_set_header_on_caps): Fix
- gst_value_list calls on arrays. Fixes #321962.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: fix a small glitch after pause
+ After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
+ the amount of output samples we consumed. We can't do this reliably with the
+ current API when we are doing trick modes but we can do the right thing for
+ normal playback.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
- * gst/adder/gstadder.c: (gst_adder_init),
- (gst_adder_request_new_pad), (gst_adder_collected),
- (gst_adder_change_state):
- Update for gst_collectpads_foo() to gst_collect_pads_foo()
- API change.
+ * gst/playback/gstplaysink.c:
+ playbin2: better error message on sink failure
+ If we could create the sinks, but the don't work, don't send the missing plugin
+ message and report that the state-changed failed.
-2005-11-21 Michael Smith <msmith@fluendo.com>
+2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
- (gst_vorbisenc_chain), (gst_vorbisenc_output_buffers):
- Properly handle pad_push return values.
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: don't leak pad-template
+ gst_element_class_add_pad_template() does not take ownership.
-2005-11-20 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_to_vorbiscomment_buffer):
- Remove obsolete vorbistag element and debug category.
-
- * gst/playback/gstplaybasebin.c: (check_queue):
- Don't divide by 0 when queue-threshold is 0.
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_set_property):
- Don't modify an existing pixel-aspect-ratio if we fail to read
- a new one.
-
-2005-11-20 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_push_buffer),
- (gst_vorbisenc_push_packet):
- GST_PAD_IS_USABLE is gone, use the return value of
- the push or pad_alloc_buffer instead.
-
-2005-11-18 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new),
- (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_xwindow_draw_borders), (gst_ximagesink_ximage_put),
- (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_update_geometry),
- (gst_ximagesink_handle_xevents), (gst_ximagesink_event_thread),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_bufferpool_clear), (gst_ximagesink_getcaps),
- (gst_ximagesink_setcaps), (gst_ximagesink_change_state),
- (gst_ximagesink_show_frame), (gst_ximagesink_buffer_alloc),
- (gst_ximagesink_navigation_send_event),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_expose),
- (gst_ximagesink_set_property), (gst_ximagesink_get_property),
- (gst_ximagesink_finalize), (gst_ximagesink_init),
- (gst_ximagesink_class_init):
- * sys/ximage/ximagesink.h: Refactoring of ximagesink.
- This new version brings correct software scaling, non flickering
- window while resizing, pixel aspect ratio handling, usage of
- hardware buffer pools, out of data flow event thread for
- navigation and handling of expose events even when being PAUSED,
- a new property to keep video aspect ratio when resizing, etc...
-
-2005-11-18 Julien MOUTTE <julien@moutte.net>
-
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_fixate_caps): Introduce back caps fixate with
- handling of PAR.
-
-2005-11-18 Edward Hervey <bilboed@dvdsrc.fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init):
- Unsetting IS_SINK flag from the fakesink, so decodebin
- never behaves as a sink.
-
-2005-11-17 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From d0ea89e to b3941ea
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Fix the audiosrc base class again, we did not unflush.
+2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-11-17 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/interfaces/navigation.c:
+ * sys/v4l/v4lsrc_calls.c:
+ navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
- * examples/seeking/seek.c: (make_dv_pipeline),
- (make_vorbis_theora_pipeline), (make_avi_pipeline),
- (make_mpegnt_pipeline): Use VSINK everywhere, add a scaler
- to ogg/vorbis/theora pipeline.
+2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-11-17 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoradec.c:
+ theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
+ This fixes most seeking issues when used with gnonlin.
+ Fixes #543591
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_collected), (gst_ogg_mux_change_state):
- Fix EOS on multiple streams.
- More debugging.
+2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From f8b3d91 to d0ea89e
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_configure_segment),
- (gst_ogg_demux_perform_seek):
- Segment done must include stream time.
+2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_clear), (gst_ogg_mux_init), (gst_ogg_mux_finalize),
- (gst_ogg_mux_change_state):
- Fix ogg muxer again.
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't leak selector when getting current stream numbers.
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init):
- Fix compile again.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: use fully qualified urls when using a proxy
+ Use a fully qualified url when specifying the url for tunneled requests through
+ a proxy.
+ See #573173
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/libvisual/visual.c: (gst_visual_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_init),
- (gst_ogg_parse_chain):
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init):
- * ext/theora/theoradec.c: (gst_theora_dec_init):
- * ext/theora/theoraenc.c: (gst_theora_enc_init):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init):
- * gst/adder/gstadder.c: (gst_adder_class_init),
- (gst_adder_dispose):
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_init):
- * gst/subparse/gstsubparse.c: (gst_subparse_init):
- * gst/videorate/gstvideorate.c: (gst_videorate_init):
- Fix a whole set of pad template leaks
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/navigation.c:
+ * gst-libs/gst/interfaces/navigation.h:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.gitignore:
+ * tests/check/libs/navigation.c:
+ * win32/common/libgstinterfaces.def:
+ navigation: Extend the navigation interface
+ Add support for a set of standard commands that can be queried and executed to
+ support applications like DVD. Add query construction and parsing functions.
+ Add new messages that can be sent on the bus to provide notifications related
+ to commands, multiangle changes, and button highlight activity.
+ Add some helper functions to parse the existing GstNavigation events that
+ elements might receive.
+ Document it all and add unit tests.
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
- * check/generic/states.c: (GST_START_TEST):
- fix the test so that it only checks for elements that are part of
- this source module
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybasebin.h:
+ playbin: Add simple 'raw decoding mode'.
+ Raw decoding mode removes almost all buffering in video and audio queues
+ when a source providing already decoded video/audio is detected, on the
+ possibly bogus assumption that such a source should provide sufficient
+ internal queueing. Fixes playback on some DVDs, and improves it
+ on all.
-2005-11-16 Michael Smith <msmith@fluendo.com>
+2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear), (gst_ogg_mux_reset),
- (gst_ogg_mux_init), (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
- (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected), (gst_ogg_mux_clear_collectpads),
- (gst_ogg_mux_change_state):
- Fix leaking collectpads.
+ * tests/check/elements/.gitignore:
+ ignores: Ignore the videoscale check binary
-2005-11-16 Edward Hervey <edward@fluendo.com>
+2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_event), (gst_videorate_chain):
- Handle segment seeks
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_set_proxy to the win32 exports
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock),
- (gst_base_audio_sink_change_state):
- Set ringbuffer to non-flushing when going to PAUSED, set to
- flushing again when going to READY.
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: don't forget to release locks in a few places
+ Might fix #576585.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
- (gst_ring_buffer_stop):
- Start in flushing mode by default.
- Don't set flushing in the _stop method, let the app call
- this explicitly.
+2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Julien MOUTTE <julien@moutte.net>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't read over line ends when taking the last Cr or Cb
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_center_rect):
- * gst-libs/gst/video/videosink.h: Add helper function needed
- for video sinks.
+2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't write to few pixels and don't mix Cr and Cb
+ Fixes bug #577054.
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- Don't leak reference to pad parent.
+2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ * gst/audioresample/gstaudioresample.c:
+ * tests/check/elements/audioresample.c:
+ audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
+ If one side has a preference for a particular sample rate or set of sample rates, we
+ should honour this in the caps we advertise and transform to and from, so that elements
+ actually know about the other side's sample rate preference and can negotiate to it
+ if supported. Also add unit test for this.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop):
- Set ringbuffer to flushing when stopping so that we don't
- block on wait_segment anymore and livelock.
+2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ docs: add a blurb about redirect messages to playbin2 docs
- * examples/seeking/seek.c: (send_event), (do_seek),
- (loop_toggle_cb), (segment_done), (main):
- Added looping checkbox.
+2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix little typo in the comments
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_init):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_base_init),
- (gst_vorbis_dec_init):
- revert unrefs, they don't pass make check
+2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-15 Johan Dahlin <johan@gnome.org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
+ People might queue messages from a thread other than the thread in which
+ the main context which this watch is attached is iterated from, so use
+ a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
+ over list nodes just freed in the other thread. This just fixes issues
+ I've had with gst-rtsp-server. We might need more locking in various
+ places here.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_init), (gst_ogg_demux_finalize):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_base_init),
- (gst_vorbis_dec_init):
- Fix pad template leaks.
+2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: clear the entire builder structure
+ And use structure instead of variable with sizeof when
+ clearing the rtsp message structure, for clarity.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- Make state change function thread safe.
+2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-15 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ docs: fix typo in gst_rtsp_message_unset() API docs
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
- (gst_ogg_demux_class_init):
- Implement GstElement::send_event, so we can send seek events
- in GST_STATE_READY
+2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-14 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: add support for proxies
+ Add suport for proxy servers. Currently only used for tunneled HTTP
+ connections without authentication.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new):
- Discovered how to take away flickering while resizing the
- window. Please don't put that in ximagesink, refactoring in
- progress.
+2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-14 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
+ This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_client_queue_data),
- (gst_multifdsink_render):
- Don't leak GDP headers when using GDP mode (i.e. tcpserversink).
+2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-11-14 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
+ According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
+ format the colorkey depending on xcontext->depth. This is what they will use to
+ interprete the value. The max_value in turn is usualy a constant regardless of
+ the depth.
- * gst/playback/gstplaybin.c: (gen_audio_element):
- Use autoaudiosink, it tends to be more widely available than
- autoaudiiosink.
-
-2005-11-14 Andy Wingo <wingo@pobox.com>
+2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c (gen_audio_element): Use autoaudiosink
- as well if it is available. Fixes #316442.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: reset whole message (was sizeof pointer instead of sizeof type)
-2005-11-14 Michael Smith <msmith@fluendo.com>
+2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear), (gst_ogg_mux_init),
- (gst_ogg_mux_sinkconnect), (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
- (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_collected),
- (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_buffer_from_packet),
- (gst_vorbisenc_change_state):
- Fix a small memory leak in vorbisenc.
- Fix large memory leaks in oggmux, also fix lots of state change
- bugs in oggmux.
+ * gst-libs/gst/interfaces/mixer.c:
+ doc: Fix a typo in the GstMixer docs
-2005-11-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_init),
- (gst_videotestsrc_src_fixate):
- move fixation to a fixate function
- remove negotiate function, basesrc's is good enough
- fixes a bug for check when using the element alone
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix linear scaling for one byte components
+ Fixes bug #577054.
-2005-11-13 Edward Hervey <edward@fluendo.com>
+2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * examples/seeking/seek.c: (do_seek), (accurate_toggle_cb),
- (key_toggle_cb), (main):
- Added checkboxes for adding/removing the accurate and key_unit seek
- flags.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Fix 4tap scaling of YUYV and friends
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac: back to HEAD
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
+ Partially fixes bug #577054, there's just one issue left now.
-=== release 0.9.5 ===
+2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * tests/check/elements/videoscale.c:
+ videoscale: Add some more unit tests
- * configure.ac:
- releasing 0.9.5, "No No Kia"
+2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-11 Edward Hervey <edward@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use bilinear instead of 4tap scaling for heights < 4
+ Partially fixes bug #577054.
- * examples/seeking/seek.c: (make_parselaunch_pipeline):
- Added parse-launch syntax seeking mode for the seeking example.
- This should help stress-test even more cases.
- Ex usage : ./seek 15 "filesrc location=uranus.avi ! decodebin ! xvimagesink"
+2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
+ This case is for upscaling a frame with width=1
+ Partially fixes bug #577054.
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Check whether peer pad exists before sending navigation events
- to it.
+2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-11 Michael Smith <msmith@fluendo.com>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
+ Partially fixes bug #577054.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_setup),
- (gst_vorbisenc_buffer_from_packet):
- * ext/vorbis/vorbisenc.h:
- Set duration on encoded buffers. This allows oggmux's
- max_page_delay parameter to actually work.
+2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Initialize buffer memory with zeroes
+ This prevents valgrind warnings when accessing the "x" parts
+ of xRGB and friends in other elements that handle (and can handle)
+ xRGB like ARGB (for example videoscale).
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_get_palette), (gst_ffmpeg_set_palette),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size), (gst_ffmpegcsp_transform):
- Make palettes work again (see #132341). Use our own macros
- for rounding up.
+2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-10 Andy Wingo <wingo@pobox.com>
+ * tests/check/Makefile.am:
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a lot of unit tests
- * gst/playback/gstplaybasebin.c (prepare_output): Fix format
- string doober.
+2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videoscale/gstvideoscale.c:
+ videocale: Add support for video/x-raw-gray with bpp=depth=8
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps):
- Prefer passthrough in transform_caps
+2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- check for ALSA errors properly, instead of relying on ALSA's
- error strings to serve to the user.
+2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-10 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_event), (gst_ogg_demux_init),
- (gst_ogg_demux_configure_segment), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
- Modernise the seek code.
+2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-10 Michael Smith <msmith@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (group_commit), (new_decoded_pad),
- (setup_substreams), (set_active_source):
- Unlock GROUP_LOCK in failure cases, so that we don't deadlock when
- trying to go to NULL if we failed to read a file.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
-2005-11-10 Wim Taymans <wim@fluendo.com>
+2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiotestsrc_class_init), (gst_audiotestsrc_get_times),
- (gst_audiotestsrc_create):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_get_times), (gst_sinesrc_create):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_get_times),
- (gst_videotestsrc_create):
- The base class can now sync for us.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for v308 YUV colorspace
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate): Check if the caps have
- a fourcc field. Fixes crash for gst-launch-0.9 v4lsrc
- name=source autoprobe=false autoprobe-fps=false copy-mode=1
- device=/dev/video0 ! ffmpegcolorspace ! "video/x-raw-yuv,
- format=(fourcc)I420" ! xvimagesink
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Add my copyright to the 4tap scalers
-2005-11-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_get_query_types), (gst_sinesrc_src_query),
- (gst_sinesrc_newsegment):
- Send newsegment event in TIME format, set duration if
- num-buffers is set, fix duration querying.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Enable 4-tap scaling for all supported formats
-2005-11-10 Michael Smith <msmith@fluendo.com>
+2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
- (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected):
- Fix EOS handling, partially. Now forwarding an EOS event once we have
- EOS on all pads works correctly. However, we still don't properly set
- EOS on the actual ogg stream pages.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB565 and RGB555
-2005-11-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose):
- Set elements to NULL state before disposing of them.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for UYVY
-2005-11-08 Sebastien Cote <sebas642 at yahoo dot ca>
+2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for YUY2 and YVYU
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_set_gst_timestamp):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- We need to send a newsegment event for each instance, not
- just for the first instance of this class (get rid of
- static variable in function). (#321011).
-
-2005-11-08 Michael Smith <msmith@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_push_buffer),
- (gst_ogg_mux_dequeue_page), (gst_ogg_mux_pad_queue_page),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
- Forward port rewrite of muxing strategy to 0.9 version of oggmux.
- This makes us mux things correctly according to the ogg muxing
- rules. Still not handling EOS correctly right now, though.
-
-2005-11-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Fix typo in docs.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB and BGR
-2005-11-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
- (gst_ogg_pad_submit_packet), (gst_ogg_chain_new):
- Initialise segment_stop to GST_CLOCK_TIME_NONE when
- creating a new chain; should fix live streaming. Also
- add more debug output and fix a typo.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
-2005-11-08 Brian Cameron <brian dot cameron at sun dot com>
+2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Fix drawing of UYVY text borders
- * gst/volume/gstvolume.c: (volume_set_caps):
- Fix compilation on Solaris with Forte. (#320923)
+2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
-2005-11-08 Wim Taymans <wim@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Add support for UYVY colorspace
+ Fixes bug #378094.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- No need to do a typecheck.
+2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: do some more cleanup
+ Free the groups when we go to READY.
+ Allow for NO_PREROLL elements.
- * ext/alsa/gstalsa.h:
- We register a debug category, so let's use it.
+2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-04 Zeeshan Ali <zeenix@gmail.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: start CSeq counting from 1 instead of 0
+ Start counting from 1 instead of 0 as this is what most other clients
+ seem to do.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_wait):
- Fixed a small problem.
+2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-04 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add ETag and If-Match headers
+ Add new headers, we need them for RealMedia support.
- * examples/seeking/Makefile.am:
- * examples/seeking/scrubby.c: (gst_element_factory_make_or_warn),
- (dynamic_link), (setup_dynamic_link), (make_wav_pipeline),
- (make_playerbin_pipeline), (format_value), (update_scale),
- (speed_cb), (seek_cb), (do_seek), (start_seek), (stop_seek),
- (play_cb), (pause_cb), (stop_cb), (print_message), (bus_message),
- (print_usage), (main):
- Added app for playback speed testing.
+2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
- * examples/seeking/seek.c: (dynamic_link),
- (make_avi_msmpeg4v3_mp3_pipeline), (make_avi_pipeline),
- (make_mpeg_pipeline), (do_seek), (set_update_scale),
- (message_received), (main):
- Updated seek example.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: scale the colorkey components in case of 16bit visuals
+ Use a default that won't be scales to 0,0,0
-2005-11-04 Zeeshan Ali <zeenix@gmail.com>
+2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_thread), (gst_base_rtp_depayload_wait),
- (gst_base_rtp_depayload_set_clock):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Don't sleep on the bench (system clock) when you have a nice
- comfortable bed (Gstreamer clock) to sleep on.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audiosrc: improve 'Dropped n samples' warning message
-2005-11-03 Wim Taymans <wim@fluendo.com>
+2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (gst_decode_bin_dispose), (free_dynamics), (remove_fakesink),
- (pad_blocked), (close_pad_link), (new_pad), (no_more_pads):
- Handle the case where a pad_block failed.
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ examples: use new method to set flags
+ Use the new core method for setting object enum properties by name.
-2005-11-02 Sebastien Cote <sebas642@yahoo.ca>
+2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: add more support for subpictures
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- Fixes some bugs in the depayloader's queuing/de-queueing code.
-
-2005-10-31 Michael Smith <msmith@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_print):
- Patch from Alessandro Decina <alessandro@nnva.org>.
- Make oggdemux only find the final time in a chain, not per-pad,
- since the per-pad information can be very expensive to locate, and
- it isn't used anywhere. This makes reading a file containing
- OggSkeleton reasonably fast.
- Also, make chain finding work when there are logical bitstreams that
- can't be decoded. Fixes #319110.
-
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release),
- (gst_base_rtp_depayload_start_thread),
- (gst_base_rtp_depayload_set_property),
- (gst_base_rtp_depayload_get_property):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Some random fixes, to fullfill the desires of thomas.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: first support for subpictures
+ Add beginnings of subpicture support.
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push):
- Fixed the queueing algorithm.
+ * tests/examples/seek/seek.c:
+ seek: print tags from the different tracks
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_push):
- A small fix
+ * gst/playback/gstplaybin2.c:
+ playbin2: blacklist subpictures for now
+ Blacklist the subpictures until we add support for them.
+ Add some small debug info.
+ See #576408.
-2005-10-31 Wim Taymans <wim@fluendo.com>
+2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.h:
- Don't break ABI.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: expose more media types
+ Expose more media types from a raw source, such as the subpicture and various
+ text pads.
+ Small cleanups and add some more debugging.
+ See #576408.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_set_caps):
- Some more comments.
- Handle missing required caps fields better.
+2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-31 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playbin2: rescan audio sinks for volume/mute
+ Rescan the audio sinks for the mute and volume properties.
+ fixes #576180.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
- (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
- (gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add flushing mode to the ringbuffer so that it in all cases does
- not try to handle more audio. This makes sure it does not try to
- block anymore when flushing and fixes a livelock.
+2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playbin2: fix reuse of the video chains
+ When reusing playbin with visualisations, reset the async property on the video
+ sink because some sinks might dynamically recreate their sinks.
+ Fixes #576188
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_convert),
- (gst_ogg_demux_chain_peer), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain):
- Explicitly check for -1 values before doing a conversion
- and always map them to -1. (#315545)
+2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaysink.c:
+ playbin2: allow dynamic swtiching of subtitles
+ When we have the textpad configured, enable and disable the subtitles by setting
+ the silent flag on the overlay element instead of trying to remove elements.
+ See #576187
- * gst/playback/gstplaybin.c: (gen_video_element):
- first try autovideosink, then xvimagesink, then error out
- * po/POTFILES.in:
- add translatable file
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- update translations
+2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-27 Philippe Khalaf <burger at speedy dot org>
+ * tests/icles/playbin-text.c:
+ tests: print some more info in the text example
+ Print both the position and the running_time when the subtitle becomes available
+ in the application.
- * gst-libs/gst/rtp/gstbasedepayload.c:
- * gst-libs/gst/rtp/gstbasedepayload.h:
- Minor cleanups
+2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playbin2: fix dynamic switching of visualisations
+ Fix the switching of visualisations by requesting and releasing the tee request
+ pads on demand.
+ See #576187.
- * gst/playback/.cvsignore:
- * gst/playback/decodetest.c:
- * gst/playback/test3.c:
- Port these two tests as well.
+2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-27 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/README:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ docs: add examples for tcp elements, also use correct section name. Fixes #564139
+ Updated the examples in the README to actually work. Add them to api docs. Tests
+ the api-docs and fix the section names to make the docs actualy show up.
+ The example for "tcpserversrc" needs review (might be an element bug).
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/theora/theoraenc.c: (theora_enc_sink_event),
- (theora_enc_change_state):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_change_state):
- Take proper locks when handling events.
+2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-27 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ indent: fix damange that gst-indent did some time ago
- * gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
- (gst_adder_change_state):
- Fix timestamps and fix deadlock when stopping the collectpads.
+2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-27 Zeeshan Ali <zeenix@gmail.com>
+ * gst/playback/gstplaysink.c:
+ playbin2: fix linking order
+ Link after doing the state change and unlink before shutting down. Makes the
+ window for causing races in toggling the visualisations smaller.
+ See #576187.
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Declaring the payload types as strings too so that they can be used
- in the padtemplate inialization.
+2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-26 Andre Magalhaes <andrunko@gmail.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: reset counter
+ reset the number of pending dynamic operations back to 0 when we reuse
+ uridecodebin.
+ Fixes #576190
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init):
- Fixes a small but nasty bug. The derived elements no longer segfaults
- on finalization.
+ * ext/theora/theoradec.c:
+ theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
+ The problem was that previously we didn't check whether _theora_granule_frame
+ returned a negative framecount or not, resulting in bogus timestamps.
-2005-10-26 Michael Smith <msmith@fluendo.com>
+2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
- * gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
- When clearing an audioconvert context, set tmpbufsize to zero, so
- we'll allocate it again later if required.
- This fixes audioconvert re-negotiating formats, which previously
- segfaulted with a NULL destination buffer.
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Set caps on non-header ouput buffers.
+ Fixes #576142.
-2005-10-26 Zeeshan Ali <zeenix@gmail.com>
+2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_base_init), (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- Fixed a smalll memleak.
+ * tests/examples/seek/seek.c:
+ seek: Add some more debug
+ Add some more info about the selected streams.
-2005-10-26 Zeeshan Ali <zeenix@gmail.com>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_base_init), (gst_base_rtp_depayload_init),
- (gst_base_rtp_depayload_finalize),
- (gst_base_rtp_depayload_setcaps),
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release),
- (gst_base_rtp_depayload_thread),
- (gst_base_rtp_depayload_change_state):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Changed the C++ comments to C comments
+2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-26 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: a pad starts out being not drained.
+ Mark a new pad as not drained until we get EOS on it.
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init):
- * gst/tcp/gsttcpserversrc.h:
- Remove unused 'curoffset' structure member.
+2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
-2005-10-25 Zeeshan Ali <zeenix@gmail.com>
+ * gst/playback/gstqueue2.c:
+ win32: fix seeking in large files
+ Fix Seeking in large files by using the 64-bit seek functions.
+ Fixes #576019
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_base_init),
- (gst_base_rtp_depayload_finalize):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.h:
- The pad-template on the sinkpad should be set by the derived classes.
- Also added some useful macros.
+2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: recover from failing to add a pad
+ When we cannot add a pad to the decodebin2 for some reason, print a warning but
+ continue adding the remaining pads.
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Correctly flush decoder samples even if we could not
- copy them to an output buffer. Fixes #319618.
+2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-24 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: more cleanups and docs.
+ Add some more comments and use g_list_prepend().
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Check
- the caps against our xcontext caps.
+2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: refactoring and race fixes
+ Refactor some code so that we can take the right locks and in the right order.
+ Fixes quite a bit of races already.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
- Remove g_print
- Use sync property from baseclass to disable sync.
+2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: remove the group cond + cleanups
+ Remove the group GCond that we used for waiting for groups to finish because we
+ use pad blocking on the selectors and counters instead for waiting for the
+ groups to complete.
+ remove the obsolete about_to_finish variable set while emiting the
+ about-to-finish signal and fix some old comments.
+ We don't need to take the playbin lock when querying the uridecodebin.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
- Buffers with no timestamps get aligned with previous buffers or
- on underrun, played ASAP.
+2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-24 Julien MOUTTE <julien@moutte.net>
+ * tests/icles/playbin-text.c:
+ icles: print better error and warning messages
+ --
- * gst-libs/gst/video/video.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support): And
- here comes my change on caps for framerate and geometry range.
- We are now accepting 1 to MAXINT for width and height, and from
- 0.0 to MAXDOUBLE for framerate. That allows duration less png frames
- to be blended correctly in videomixer.
+2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-10-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspbase64.c:
+ * gst-libs/gst/rtsp/gstrtspbase64.h:
+ rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
+ This also fixes another instance of CVE-2008-4316.
- * configure.ac:
- back to HEAD
+2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.9.4 ===
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: report -1 for duration in push mode
+ In push mode we must return TRUE from the duration query with a value of -1
+ meaning that we know that we don't know the duration.
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.9.4, "Velociraptor"
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add extra dynamic ref for demuxers
+ When we make a group connected to a demuxer, keep an extra dynamic refcount for
+ the group which is only decremented when no_more_pads or a multiqueue overrun is
+ detected. This way we avoid a race between exposing the group while more dynamic
+ refs are added from new pads.
+ Fixes #575588.
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- * po/POTFILES.in:
- STOPPED -> FAILED
+ * gst/playback/gstplaysink.c:
+ playbin2: sync state of the sink correctly
+ Sync the state of the newly added chains to the state of the parent sink element
+ to avoid lost async-start messages. Fixes cdda:// async-done message storm.
-2005-10-21 Wim Taymans <wim@fluendo.com>
+2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (gst_decode_bin_dispose), (free_dynamics), (pad_unblocked),
- (pad_blocked), (close_pad_link), (new_pad):
- Don't try to remove elements twice.
+ * gst/playback/gstplaybin2.c:
+ playbin2: return NOT_LINKED for unselected streams
+ When streams are not selected in the selector, return NOT_LINKED so that
+ upstream elements can skip decoding. Only do this for audio and video pads
+ because for text streams the overhead is smaller and they could come from
+ external files.
-2005-10-21 Wim Taymans <wim@fluendo.com>
+2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisdec.h:
- Fix old naming.
+ * gst/playback/gstplaysink.c:
+ playbin: set custom text sink properties
+ Set the custom sink async=FALSE to not make it participate in preroll because we
+ are dealing with sparse streams.
+ Try to set sync=TRUE on the custom text sink.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Don't try to sync on buffers without a timestamp.
+2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-21 Wim Taymans <wim@fluendo.com>
+ * tests/icles/playbin-text.c:
+ example: use appsink instead of fakesink
+ Use appsink instead of fakesink to get the subtitles.
+ Make things more pretty.
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisdec.h:
- Fix old naming.
+2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/playbin-text.c:
+ examples: add example of intercepting subtitles
+ Add an example of how to install a custom sink for receiving subtitles in
+ playbin2.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types),
- (gst_vorbisenc_src_query):
- Implement position and duration queries.
+2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/test3.c: (update_scale), (main):
- Fix for async state changes and print nicer output.
+ * tests/check/elements/appsink.c:
+ tests: fix include in the appsink test
+ Fix dist by doing the right include.
-2005-10-20 Wim Taymans <wim@fluendo.com>
+2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- * gst/playback/test6.c: (main):
- Fix tests again
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't try to set invalid stream numbers
+ Fix a problem with setting the stream numbers because we check for the wrong
+ range.
+ See #575239.
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
- Don't use functions for position queries when handling
- duration queries.
+ * gst/playback/gstplaybin2.c:
+ playbin2: release the shutdown lock
+ Release the shutdown lock when we wait for other groups to complete or else we
+ have a deadlock when the other group completes and tries to grab the shutdown
+ lock.
+ Fixes #575550.
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_data_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- * ext/vorbis/vorbisdec.h:
- Vorbis streams can be embedded in other container formats
- than ogg, container formats where the demuxer might set
- timestamps on encoded vorbis buffers instead of those silly
- granulepos thingies. In short: make vorbisdec handle
- timestamps on incoming buffers as well.
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ * tests/examples/app/appsrc-stream.c:
+ * tests/examples/app/appsrc-stream2.c:
+ examples: fix g_object_set() value type.
+ Make sure we cast the length value as a gint64 to the vararg g_object_set() just
+ incase sizeof(gsize) != sizeof(gint64).
-2005-10-20 Wim Taymans <wim@fluendo.com>
+2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gst_play_base_bin_change_state):
- Fix leak.
- Handle case where playbasebin is now ASYNC because
- decodebin is.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: make flac typefinder return lower probability for frame headers
+ The flac frame header typefinder overstates the likelihood of a match, leading
+ to false positives with e.g. aac streams and PDF files. Reduce probabilty
+ returned from LIKELY to POSSIBLE for the frame header matchin code.
+ Fixes #574939.
-2005-10-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/bufferframesconvert.c:
- * gst/audioconvert/plugin.c: (plugin_init):
- * gst/audioconvert/plugin.h:
- And bye bye buffer-frames-convert
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: improve image/bmp typefinder
+ Detect more variations and also bail out in more cases where the values
+ don't make sense. Furthermore, add width/height and bpp to the caps,
+ because we can.
-2005-10-19 Wim Taymans <wim@fluendo.com>
+2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
- * check/elements/audioconvert.c:
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_identification_packet), (vorbis_handle_data_packet):
- * ext/vorbis/vorbisenc.c: (raw_caps_factory):
- * gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
- * gst-libs/gst/audio/audio.h:
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps):
- * gst/volume/gstvolume.c:
- Bye bye buffer-frames.
-
-2005-10-19 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
- (query_positions_elems), (query_positions_pads), (update_scale),
- (do_seek), (set_update_scale), (message_received), (main):
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_find_chains),
- (gst_ogg_demux_loop):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_data_packet):
- * gst/adder/gstadder.c: (gst_adder_query):
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiotestsrc_src_query):
- * gst/playback/test3.c: (update_scale):
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- * gst/playback/test6.c: (main):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
- Query API update.
-
-2005-10-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
- (xml_check_first_element), (xml_type_find), (smil_type_find),
- (plugin_init):
- Add typefinding for SMIL and for generic XML. Based on patch by
- Akos Maroy (#308663).
-
-2005-10-18 Wim Taymans <wim@fluendo.com>
+ * tests/check/Makefile.am:
+ check: Ignore alsamixer in the states test too
- * gst/playback/Makefile.am:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (gst_decode_bin_dispose), (dynamic_create), (dynamic_free),
- (free_dynamics), (pad_unblocked), (pad_blocked), (close_pad_link),
- (try_to_link_1), (new_pad), (no_more_pads), (type_found),
- (gst_decode_bin_change_state):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_send_event_to_sink):
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- * gst/playback/test6.c: (main):
- Make playbin async, it'll commit state to paused when all streams
- are detected.
- Remove ugly hack.
- Added test6.c to show async behaviour.
-
-2005-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_loop):
- Fix for segment-start/stop API change.
-
-2005-10-18 Wim Taymans <wim@fluendo.com>
-
- * check/Makefile.am:
- * check/clocks/selection.c: (GST_START_TEST), (volume_suite),
- (main):
- Add future test for clock selection.
-
-2005-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsasink.c: (gst_alsasink_init),
- (gst_alsasink_close):
- Set handle to NULL.
+2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
- (gst_ring_buffer_acquire), (gst_ring_buffer_release),
- (gst_ring_buffer_start), (gst_ring_buffer_pause),
- (gst_ring_buffer_stop), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- More debug info.
+ * sys/v4l/v4l_calls.c:
+ v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
-2005-10-17 Stefan Kost <ensonic@users.sf.net>
+2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/Makefile.am:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- fix broken build of controllerized plugins
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix resolving of hostnames
+ We were returning a pointer to a stack variable with the resolved hostname,
+ which doesn't work.
+ return a copy of the resolved ip address instead.
+ Fixes #575256.
-2005-10-17 Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisparse.c:
+ vorbisparse: be smarter when queueing headers
+ Look at the first buffer byte to see if a buffer is a header instead of counting
+ packets.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for Indeo-3 (IV32).
+2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c:
+ theoraparse: be smarter when queuing headers
+ Look at the first byte of the buffer data (if we can) to decide if the packet is
+ a header packet or not instead of counting packets.
- * configure.ac:
- rewrite
+2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add some debug info
+ Add some debug info to log when the seek worked.
- * gst-libs/gst/video/video.c: (gst_video_get_size):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- doc updates
+2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-17 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: release lock in _eos flushing case
+ Release the mutex when we are flushing in gst_app_src_end_of_stream()
+ Fixes #574964.
- * ext/ogg/gstoggmux.c (gst_ogg_mux_queue_pads): Fix bug introduced
- with the collectpads change.
- (gst_ogg_mux_send_headers): Elevate warning to a g_critical.
+2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/tcp/gstmultifdsink.c: Convert to use the boilerplate macro.
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
- * gst/tcp/gsttcp.c (gst_tcp_socket_read): Comment update.
+2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
- * ext/theora/theoraenc.c (theora_buffer_from_packet): Pass the
- alloc_buffer flow return to callers.
- (theora_enc_chain, theora_enc_chain): Adapt to buffer_from_packet
- change. Fix some memleaks in theoraenc.
+ * ext/theora/theoradec.c:
+ theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
- * ext/ogg/gstoggmux.c (gst_ogg_mux_send_headers): Fix a segfault
- in strange circumstance.
+2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-17 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gsturidecodebin.c:
+ playbin2: fix raw elements like cdda://
+ Fix a fixme with a one liner and make cd playback work again.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size): We are asked to compute a buffer size
- from caps, let's use the caps...
+2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: improve subtitle handling
+ Add property to playbin2 to configure a custom sink that receives the raw
+ subtitle buffers instead of using a textoverlay.
+ Improve the property finding code to make it more usable.
+ Use property find code to find async properties in custom sinks that are bins.
+ Improve text overlay code to gracefully handle missing elements.
- * configure.ac:
- put back AX_CREATE_STDINT_H, ffmpegcolorspace includes _stdint.h
+2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Protect memory allocation calculation from overflow.
+ Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
- * gst-libs/gst/interfaces/Makefile.am:
- fix silly typo
+2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
-2005-10-16 Andy Wingo <wingo@pobox.com>
+ * gst-plugins-base.spec.in:
+ Spec: fix up deps
- * gst/playback/gstdecodebin.c
- (gst_element_set_state_like_a_crazy_man): New kraaaaaaazy
- function!
- (try_to_link_1): Increase kraziness level.
+2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix parsing of the timeout parameter
+ --
- * configure.ac:
- restructure like the core one
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- use correct linker flags, now the libs are properly versioned
- * check/elements/audioconvert.c: (verify_convert):
- * ext/alsa/gstalsaplugin.c:
- * ext/cdparanoia/gstcdparanoia.c:
- * ext/gnomevfs/gstgnomevfs.c:
- * ext/libvisual/visual.c:
- * ext/ogg/gstogg.c:
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_chain):
- * ext/theora/theora.c:
- * ext/vorbis/vorbis.c:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- * gst-libs/gst/tag/gsttagediting.c:
- * gst-libs/gst/video/video.c:
- * gst/adder/gstadder.c:
- * gst/audioconvert/plugin.c:
- * gst/audiorate/gstaudiorate.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/resample_ref.c: (resample_scale_ref):
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/resample.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/ffmpegcolorspace/gstffmpeg.c:
- * gst/playback/gstdecodebin.c: (close_pad_link):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- * gst/sine/gstsinesrc.c:
- * gst/subparse/gstsubparse.c:
- * gst/tags/gsttagediting.c:
- * gst/tcp/gsttcpplugin.c:
- * gst/typefind/gsttypefindfunctions.c:
- * gst/videorate/gstvideorate.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- * sys/v4l/gstv4l.c:
- * sys/ximage/ximage.c:
- * sys/xvimage/xvimagesink.c:
- fix up defines
+2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-13 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: fix g_return condition
+ when parsing a data message, we require a data message.
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
- (gst_tag_to_vorbis_comments):
- Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
+2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-13 Stefan Kost <ensonic@users.sf.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: flac typefinder fixes
+ Use scan context for initial peek as well. Peek 6 bytes in the initial
+ peek rather than 5 bytes, to match the length of the memcmp we're doing
+ on that data later. Return immediately when we found caps from looking
+ at the beginning of the data - no point in continuing to scan the next
+ 64kB for something matching a frame header.
- * examples/stats/mp2ogg.c:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- typo fixes
+2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-13 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: free the right string.
+ Free the key value before we remove the header item from the array. The item we
+ retrieved from the array is only valid until we remove it from the array.
- * ext/ogg/gstoggmux.c:
- Use magic glib macros to define constants as 64 bit, to ensure
- appropriate vararg passing.
+2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-13 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: keep track of amount of decoded bytes
+ Keep track of the actual amount of decoded bytes, which can be less than 3 when
+ we decode the last bits of a base64 message.
- * ext/ogg/gstoggmux.c:
- * gst/audioconvert/audioconvert.c: (float):
- Don't use LL suffix, as it's not portable, and neither of these
- uses required it anyway.
+2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-12 Stefan Kost <ensonic@users.sf.net>
+ * gst/adder/gstadder.c:
+ adder: log details in getcaps like in setcaps
- * examples/indexing/indexmpeg.c: (main):
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get),
- (cdparanoia_open), (cdparanoia_close), (cdparanoia_event),
- (cdparanoia_convert), (cdparanoia_query):
- * ext/cdparanoia/gstcdparanoia.h:
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_init),
- (gst_gnomevfssink_open_file), (gst_gnomevfssink_close_file),
- (gst_gnomevfssink_chain), (gst_gnomevfssink_change_state):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_init):
- * gst/playback/gststreamselector.c: (gst_stream_selector_init):
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_init),
- (gst_multifdsink_render), (gst_multifdsink_start),
- (gst_multifdsink_stop):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init),
- (gst_tcpclientsink_render), (gst_tcpclientsink_start),
- (gst_tcpclientsink_stop):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init),
- (gst_tcpclientsrc_getcaps), (gst_tcpclientsrc_create),
- (gst_tcpclientsrc_start), (gst_tcpclientsrc_stop):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init),
- (gst_tcpserversrc_create), (gst_tcpserversrc_start),
- (gst_tcpserversrc_stop):
- * gst/tcp/gsttcpserversrc.h:
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init):
- * sys/ximage/ximagesink.c: (gst_ximagesink_init):
- renamed GST_FLAGS macros to GST_OBJECT_FLAGS
- moved bitshift from macro to enum definition
+2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-12 Wim Taymans <wim@fluendo.com>
+ * win32/MANIFEST:
+ win32: update MANIFEST, fixing 'make dist'
- * examples/seeking/Makefile.am:
- Oops.
+2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-10-12 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 7032163 to f8b3d91
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read), (gst_ring_buffer_clear):
- Don't assert on normal stuff.
+2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
- * gst/playback/gstplaybin.c: (do_playbin_seek):
- API fix.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add photoshop typefind functions
+ Add photoshop typefind functions.
+ Fixes #574516.
-2005-10-12 Wim Taymans <wim@fluendo.com>
+2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- * examples/seeking/Makefile.am:
- * examples/seeking/cdplayer.c: (play_cb), (pause_cb), (stop_cb):
- * examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
- (do_seek), (set_update_scale), (message_received), (main):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read), (gst_ring_buffer_clear):
- Update for _get_state() API change.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: only remove pads that were added
+ Flag pads that were added so that we can see if we need to remove them later or
+ not.
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Cleanups.
- Commit and read from ringbuffer in samples rather than bytes.
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsp: only add ports when not using TCP
+ Only add the port numbers in the transport string when we are using udp or
+ multicast.
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Respect segment rate and accum when scheduling samples.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: use gstreamer dump mem
+ --
-2005-10-11 Julien MOUTTE <julien@moutte.net>
+2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected): Quick hack to fix build. We need to handle
- EOS correctly, that needs more work.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: use glib base64 encoder
+ --
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * check/generic/states.c: (GST_START_TEST):
- remove old property.
+ * gst/playback/gstdecodebin2.c:
+ Unblock blocked ghostpads when shutting down. Fixes #574293.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek):
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_data_packet):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- * gst/videorate/gstvideorate.c: (gst_videorate_event):
- Update for newsegment API change.
+2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-11 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add mapping for Fraps video codec.
+ Found through insanity testrun. Confirmed mapping in libavformat.
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
- (do_playbin_seek), (gst_play_bin_send_event):
- Override send_event differently, so that we can takes bits of
- functionality from GstPipeline (special handling for seeks,
- including pausing/resuming, and resetting stream time) and still get
- the appropriate behaviour of only forwarding event to a single sink,
- rather than all of them.
- Unfortunately requires a lot of code duplication, but the
- alternatives are equally ugly in the end.
+2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add the 'DVR ' mapping for mpeg2video.
+ Found this in 3 files from the insanity suite and mapping is also present
+ in libavformat.
- * check/elements/audioconvert.c: (setup_audioconvert),
- (cleanup_audioconvert), (get_int_caps), (verify_convert),
- (GST_START_TEST), (audioconvert_suite):
- clean up tests a little, fix some leaks.
+2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Use the proper data pointer instead of poking random memory.
- * ext/alsa/gstalsasink.c:
- Also allow unsigned int.
+2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Small cleanup
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix compilation on windows.
+ Remove unused variable when building for windows.
+ Fixes #574443.
-2005-10-10 Wim Taymans <wim@fluendo.com>
+2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Small update, use API as stated in design docs.
+ * common:
+ Automatic update of common submodule
+ From ffa738d to 7032163
- * examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
- (update_scale), (do_seek), (seek_cb), (set_update_scale),
- (start_seek), (stop_seek), (play_cb), (pause_cb), (stop_cb),
- (message_received), (main):
- Updated seek example for GOption. Some usability improvements.
+2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 3f13e4e to ffa738d
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
- (gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
- Alloc temp storage somewhere else where we can do it more
- portable.
+2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 3c7456b to 3f13e4e
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_create),
- (gst_tcpserversrc_start):
- Don't block in accept while doing the state change, move
- to poll and make cancellable.
+2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-10-09 Philippe Khalaf <burger@speedy.org>
+ * common:
+ Automatic update of common submodule
+ From 57c83f2 to 3c7456b
- * gst-libs/gst/rtp/rtpbasedepayload.c:
- Set timestamp and add queue delay to timestamp
- * gst-libs/gst/rtp/rtpbuffer.h:
- Set correct payload type for h263
+2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoradec.c:
+ theoradec: parse and use codec_data in the caps
+ Parse the codec_data in the caps and use this as the headers.
+ Fixes #574169.
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_class_init),
- (gst_audiotestsrc_init), (gst_audiotestsrc_create_sine),
- (gst_audiotestsrc_create_square), (gst_audiotestsrc_create_saw),
- (gst_audiotestsrc_create_triangle),
- (gst_audiotestsrc_create_silence),
- (gst_audiotestsrc_create_white_noise),
- (gst_audiotestsrc_init_pink_noise),
- (gst_audiotestsrc_generate_pink_noise_value),
- (gst_audiotestsrc_create_pink_noise),
- (gst_audiotestsrc_change_wave):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- fixed typo, added pink noise
+2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add theora mapping
+ Add theora mappings. See #574169.
- * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
- (plugin_init):
- Add wavpack and spc typefind functions from 0.8 branch.
+2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * win32/common/libgstrtsp.def:
+ rtsp: Add methods for getting the read/write fds
+ API:gst_rtsp_connection_get_readfd()
+ API:gst_rtsp_connection_get_writefd()
- * gst/typefind/gsttypefindfunctions.c: (tar_type_find),
- (ar_type_find), (msdos_type_find), (plugin_init):
- Add typefind functions for tar archives, ar archives,
- RAR archives, and msdos-executables (dlls, exe, etc.).
- Some of those would be wrongly identified as mpeg
- streams of some sort before (#315550).
+2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+ * Makefile.am:
+ * win32/common/audio-enumtypes.c:
+ win32: indent copied *-enumtypes.c files in make win32-update
- * configure.ac:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
- * gst/audiotestsrc/Makefile.am:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_base_init),
- (gst_audiotestsrc_class_init), (gst_audiotestsrc_init),
- (gst_audiotestsrc_src_fixate), (gst_audiotestsrc_setcaps),
- (gst_audiotestsrc_get_query_types), (gst_audiotestsrc_src_query),
- (gst_audiotestsrc_wait), (gst_audiotestsrc_unlock),
- (gst_audiotestsrc_create_sine), (gst_audiotestsrc_create_square),
- (gst_audiotestsrc_create_saw), (gst_audiotestsrc_create_triangle),
- (gst_audiotestsrc_create_silence),
- (gst_audiotestsrc_create_white_noise),
- (gst_audiotestsrc_change_wave), (gst_audiotestsrc_create),
- (gst_audiotestsrc_set_property), (gst_audiotestsrc_get_property),
- (gst_audiotestsrc_start), (plugin_init):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- add new plugin and element
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init):
- use gobject_class
+2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/MANIFEST:
+ win32: update MANIFEST
- * gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),
- (gst_adder_init), (gst_adder_request_new_pad),
- (gst_adder_change_state):
- Add query function to source pad, so adder reports the correct
- time/sample position when queried (#315457); fix state change
- function; use GST_DEBUG_FUNCPTR() for pad functions.
+2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * win32/common/config.h:
+ win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
- Fix leaks in typefind registration
- Clean up the gratuitous commenting and whitespacing a little
+2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/multichannel-enumtypes.c:
+ * win32/common/pbutils-enumtypes.c:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ win32: update windows files via make win32-update
+ Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
+ which fixes the build of pbutils on windows (#574319).
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
- Only actually wait for the thread to be stopped if it's
- running.
+2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * .gitignore:
+ gitignore: ignore more
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- If we receive EOS we can start playback of what we had.
+2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fix build on Mac OS X
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_finalize), (multifdsink_hash_remove),
- (gst_multifdsink_stop):
- Fix crasher when going to NULL multiple times.
+2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: don't stay connected to notify::caps after negotiation
+ Disconnect the notify::caps signal in our callback (it'll be re-added
+ if we're not, in fact, finished getting complete caps). Ensures that
+ caps changes mid-stream (e.g. from an mp3 that changes from
+ stereo->mono mid-file) don't cause us to try to add a new pad.
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
- patch from Edgard Lima <edgard.lima@indt.org.br>
- Fixed gstbaseaudiosrc adding ring buffer sync to it.
+2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ rtsp: fix parsing of 'now-' ranges.
+ --
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- Report the FLOW_RETURN as string in the error message.
+2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
- Don't assert when clearing an unnegotiated buffer.
+ * tests/examples/dynamic/.gitignore:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/dynamic/sprinkle.c:
+ * tests/examples/dynamic/sprinkle2.c:
+ * tests/examples/dynamic/sprinkle3.c:
+ examples: add some more sprinkle examples
+ Add some more sprinle examples and add some more comments.
+ See #574160.
-2005-10-04 Michael Smith <msmith@fluendo.com>
+2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gen_preroll_element), (remove_groups), (setup_source):
- * gst/playback/gstplaybin.c: (remove_sinks), (add_sink),
- (setup_sinks), (gst_play_bin_send_event),
- (gst_play_bin_change_state):
- Set state to NULL before removing from bin. Fix refcounting.
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: add appsrc symbols to standard section
+ --
-2005-10-04 Michael Smith <msmith@fluendo.com>
+2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event):
- Correct refcounting in send_event() function. Previously was wrong
- if the first sink was unable to handle the event.
+ * gst/adder/gstadder.c:
+ adder: add variants for unsigned to fix warnings for unneeded check
+ For unsigned int out+in can't be < 0.
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- back to development
+ * gst/subparse/gstsubparse.c:
+ subparse: use the right variable in debug log, encoding is not yet initialized
-=== release 0.9.3 ===
+2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ * sys/v4l/v4l_calls.c:
+ v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
- * configure.ac:
- Releasing 0.9.3, "De Facto"
+2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-03 Andy Wingo <wingo@pobox.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: add missing break in event handling, remove dead code
- * gst/playback/gstdecodebin.c (try_to_link_1)
- (remove_element_chain): set element to NULL before removing it.
+2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-02 Johan Dahlin <johan@gnome.org>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: do some more cleanup in _close
+ Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
+ unconnected state as it was allocated.
- * ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_uri_get_protocols):
- protect gst_gnomevfs_get_supported_uris by a mutex, to make it
- MT safe.
+2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-02 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: fix the memory management of the url
+ Constify the url parameter in _create.
+ Make a copy of the url stored in the connection.
+ Free the url when the connection is freed.
- * gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
- (gst_ring_buffer_prepare_read):
- * gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
- Demote to LOG.
+2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-29 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * win32/common/libgstrtsp.def:
+ RTSP: Add support for server tunneling
+ Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
+ that a server can store and match the id against other tunnel requests.
+ Fix the URI in the tunnel requests so that they contain the absolute uri and the
+ query string if any instead of just the hostname.
+ Transparently base64 decode the input stream when tunneling.
+ Add method to set the connection ip address so that it can be included in the
+ tunnel response.
+ Add method to connect the two tunnel requests.
+ Add two callbacks for the async mode to notify a tunnel start and tunnel
+ complete event.
+ Add method to reset the watch after the connection has been tunneled.
+ Various little refactoring to make more stuff reusable.
+ API: RTSP::gst_rtsp_connection_set_ip()
+ API: RTSP::gst_rtsp_connection_get_tunnelid()
+ API: RTSP::gst_rtsp_connection_do_tunnel()
+ API: RTSP::gst_rtsp_watch_reset()
+
+2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer):
- * ext/theora/theoradec.c: (theora_handle_data_packet):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Propagate error codes from alloc_buffer too.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add new defines for tunneling
+ Add two more result codes for tunneling support.
-2005-09-29 Wim Taymans <wim@fluendo.com>
+2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- We use fixed caps.
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: remove , from last enum member
+ Remove , from last enum member to improve compatibility with other compilers.
- * gst/playback/Makefile.am:
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- Added example stream introspection code.
+2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
-2005-09-28 Stefan Kost <ensonic@users.sf.net>
+ * gst/subparse/gstsubparse.c:
+ subparse: Convert regex code to GRegex code
+ Fixes: #572993. Patch author prefers to use an alias, contact
+ ds if you actually need a real name.
+ Signed-off-by: David Schleef <ds@schleef.org>
- * gst/adder/gstadder.c: (gst_adder_collected):
- fix adder for float elements
+2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: remove debugging g_message
+ --
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init),
- (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init),
- (gst_base_audio_src_provide_clock):
- get_clock -> provide_clock
+2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-28 Andy Wingo <wingo@pobox.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * win32/common/libgstrtsp.def:
+ RTSP: add support for Quicktime tunneled RTSP
+ Add support for tunneling RTSP over HTTP.
+ Fix documentation some more.
+ See also #573173.
+ API: RTSP:gst_rtsp_connection_is_tunneled()
+ API: RTSP:gst_rtsp_connection_set_tunneled()
- * gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
- and unlocking.
+2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/gstsinesrc.c (gst_sinesrc_unlock): Actually implement
- unlocking.
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ RTSP: parse rtsph uris as RTSP tunneled over HTTP
+ Add transport define for RTSP tunneled over HTTP.
+ Parse rtsph:// uris as tunneled HTTP over TCP.
+ API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
+ See also #573173.
- * gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
- Actually add the pad template.
- (gst_tcpclientsink_get_type): We're a base sink. Woot, works.
+2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
- I'm at it...
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_get_url definition
+ No, I'm not wim's buildslave, seriously.
- * gst/tcp/gsttcpclientsrc.c: Make interruptable -- code stolen
- from fdsrc. Get caps in create() instead of start() so it can be
- interrupted. Interruption somewhat untested.
+2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
- Proper EOS handling.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: add _get_url method and separate sockets
+ Add gst_rtsp_connection_get_url() method.
+ Reserve space for 2 sockets, one for reading and one for writing. Use socket
+ pointers to select the read and write sockets. This should allow us to implement
+ tunneling over HTTP soon.
+ API: RTSP::gst_rtsp_connection_get_url()
-2005-09-27 Andy Wingo <wingo@pobox.com>
+2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gsttcpclientsrc.c: Cleaned up.
+ * gst-libs/gst/app/gstapp-marshal.list:
+ app: force automatic rebuild of gstapp-marshal.[ch] after previous change
+ The previous change to appsrc/appsink requires people to 'make clean'
+ to get the marshallers rebuilt (causing a build failure otherwise).
+ Change some lines in the .list file around to force a rebuild of
+ these files automatically.
- * gst/tcp/gsttcpserversrc.c: Cleaned up.
+2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
- * gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
+ * configure.ac:
+ Bump glib requirement to 2.14
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcp.c (gst_tcp_read_buffer): New function, factored
- out of tcpclientsrc.c. Cancellable.
- (gst_tcp_socket_read): Made private, cancellable, with better
- diagnostics. Also the FIONREAD ioctl takes a int*, not a size_t*.
- (gst_tcp_gdp_read_buffer): Made cancellable, actually returns the
- whole buffer, and better diagnostics.
- (gst_tcp_gdp_read_caps): Same.
+2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
- * gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
+ * ext/gio/gstgiobasesink.c:
+ gio: Use correct format modifier for size_t
+ Fixes bug #573528.
-2005-09-26 Andy Wingo <wingo@pobox.com>
+2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
- * gst/sine/gstsinesrc.h:
- * gst/sine/gstsinesrc.c: Refactor, remove the table lookup code,
- change the 'sync' property to 'is-live' and implement it halfway,
- update for controller api change.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Use correct types for some functions on Win32
+ Fixes bug #573529.
- * gst/volume/gstvolume.c (volume_transform_ip): Update for
- controller api change.
+2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-24 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix warning about using unitialized value.
- * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
- (gst_audioringbuffer_stop):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
- (gst_ring_buffer_commit), (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- Fix sync again. Moved sample alignment to basesink.
+2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add more codec mappings.
+ This comes mostly from a review of ffmpeg/libavformat/riff.c
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * gst/volume/gstvolume.c:
- add/fix docs
- * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size):
- * gst-libs/gst/audio/audio.h:
- add conversion macros for frames <-> clocktime
+2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-23 David Schleef <ds@schleef.org>
+ * ext/alsa/gstalsa.c:
+ alsa: release pcminfo after the strdup
- * gst/audioresample/Makefile.am:
- * gst/audioresample/debug.h:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/resample.c: Convert to using gst debugging
+2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ rtsprange: don't leak the range in case of parsing error.
+ Free the gstRTSPTimeRange if we don't return it. Also simplify
+ gst_rtsp_range_free() as it is valid to pass NULL to g_free().
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_send_event):
- Only seek on one sink, the first one that succeeds.
+2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-22 Michael Smith <msmith@fluendo.com>
+ * ext/alsa/gstalsa.c:
+ alsa: cleanup name lookup.
+ We can break, once we have a name to make sure, we won't read it ever twice.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_change_state):
- Don't flush encoder state unless we have an initialised encoder.
- Clear out encoder state on PAUSED_TO_READY.
+2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: don't leak line, if flushing
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_is_filled), (gst_basertppayload_push),
- (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Added max-ptime to control amount of data in the rtp packets.
+2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-21 Andy Wingo <wingo@pobox.com>
+ * ext/gio/gstgiosink.c:
+ giosink: reflow error handling to not leak uri
- * gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
- thingies.
+2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
- can be called multiple times, dogs.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: remove unused code/variables
-2005-09-21 Wim Taymans <wim@fluendo.com>
+2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_push), (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- Allow 0 ssrc too.
+ * sys/ximage/ximagesink.c:
+ ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
-2005-09-21 Stefan Kost <ensonic@users.sf.net>
+2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/compiling.sgml:
- fixing typos
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * win32/common/libgstapp.def:
+ app: add callbacks to appsrc, cleanups
+ Add a uri handler to appsink.
+ don't emit signals when we have installed callbacks on appsink.
+ Add callbacks to appsrc to replace the signals.
+ Add property to disable callbacks in appsrc, default to TRUE for backwards
+ compatibility but disable when callbacks are installed.
+ API: GstAppSrc::emit-signals
+ API: GstAppSrc::gst_app_src_set_emit_signals()
+ API: GstAppSrc::gst_app_src_get_emit_signals()
+ API: GstAppSrc::gst_app_src_set_callbacks()
+
+2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-20 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.h:
+ * tests/check/elements/appsink.c:
+ Appsink: add padding for callbacks + docs
+ Add some padding to the callbacks structure just to be safe.
+ Remove the now invisible marshaller methods from the docs.
+ Fix a comment in the unit test.
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_push), (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Added property to configure sequence number offsets.
+2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-20 Wim Taymans <wim@fluendo.com>
+ * win32/common/libgstapp.def:
+ win32: Add new libgstapp symbol
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_push), (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Make timestamp offset configurable.
+2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: clean section.txt file.
+ Add appsrc/sink symbols to private, as they are covered in the libs docs.
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- fix wrong pop/unref
+2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-09-19 Torsten Schoenfeld <kaffeetisch at gmx dot net>
+ * gst/playback/gstplaybasebin.c:
+ docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/interfaces/propertyprobe.c:
- (gst_property_probe_probe_property_name),
- (gst_property_probe_needs_probe_name),
- (gst_property_probe_get_values_name),
- (gst_property_probe_probe_and_get_values_name):
- Fix wrong macro usage; it's G_OBJECT_GET_CLASS(obj) or
- G_OBJECT_CLASS(klass), not G_OBJECT_CLASS(obj). (#316571)
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/playback/gstplaybin2.c:
+ docs: playbin2 has no stream-info
-2005-09-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
- * check/Makefile.am:
- have some tests be disabled for valgrinding
- * check/elements/vorbisdec.c: (cleanup_vorbisdec),
- (GST_START_TEST):
- * ext/vorbis/vorbisdec.c: (vorbisdec_finalize):
- Fix A Leak. Chain To Parent Finalize.
+ * gst-libs/gst/video/video.h:
+ docs: fix newly added interlace constants and plug holes in video format docs
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * examples/seeking/seek.c: (make_wav_pipeline), (main):
- Fixed wav pipeline.
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ docs: don't put random stuff in tags.
+ Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
+ tag to append text again to the documentation body.
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_push), (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- Posting ERROR and WARNING messages is good.
+ * sys/ximage/ximagesink.c:
+ ximagsink: do not access uninitialized height variable.
+ Exit like in xvimagesink, if we have partial caps.
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- This one was not supposed to go in.
+ * Makefile.am:
+ * configure.ac:
+ * win32/common/config.h.in:
+ Change how win32/common/config.h is updated
+ Generate win32/common/config.h-new directly from config.h.in,
+ using shell variables in configure and some hard-coded information.
+ Change top-level makefile so that 'make win32-update' copies the
+ generated file to win32/common/config.h, which we keep in source
+ control. It's kept in source control so that the git tree is
+ buildable from VS.
+ This change is similar to the one recently applied to GStreamer,
+ except that it adds a few -base specific defines.
+
+2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * win32/common/libgstapp.def:
+ app: add win32 .def file and only export functions we want exported
+ Add a .def file for win32 builds (and make check-exports).
+ Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
+ Make sure private marshaller functions aren't exported by prefixing them with __gst;
+ also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
+ a comment why we're not using glib-genmarshal for this one.
+
+2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/dynamic/.gitignore:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/dynamic/sprinkle.c:
+ sprinkle: Add another example app
+ Add an example app that dynamically adds and removes audiotestsrc elements from
+ adder.
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Fix for bus API.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fixed a typo.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- Some cleanups.
+2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_set_options),
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- Added debugging category.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst/tcp/gstmultifdsink.c:
+ rtsp, multifdsink: Unify the use of union gst_sockaddr.
-2005-09-18 David Schleef <ds@schleef.org>
+2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin.c: free plugin list correctly
- * gst/playback/gstplaybin.c: emit warning if autovideosink
- and autoaudiosink can't be found (instead of segfaulting)
+ * common:
+ * configure.ac:
+ build: Update shave init statement for changes in common. Bump common.
-2005-09-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/elements/audioconvert.c: (GST_START_TEST):
- try out 24 bit conversion
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ xvimageink: protect buffer_alloc from shutdown
+ Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
+ crashes when the sink is shutdown.
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_chain),
- (gst_vorbisenc_output_buffers), (gst_vorbisenc_change_state):
- * ext/vorbis/vorbisenc.h:
- Fix EOS handling. Still needs a fix in the ogg muxer to
- mark the last page as eos.
+ * gst/playback/gstplaybin2.c:
+ playbin: use flushing pads instead of fakesink
+ Use the flushing pads on playsink to terminate on shutdown instead of plugging
+ fakesinks. this should be a little cheaper.
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/tcp/gstmultifdsink.c:
- fix up ffmpegcolorspace docs; extract header
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playsink: Add FLUSHING pad type
+ Make it possible to request a flushing pad from the playsink. We can eventually
+ use these flushing pads to quickly terminate the dataflow when we are shutting
+ down.
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/theora/Makefile.am:
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c:
- * ext/vorbis/vorbisenc.c:
- pick up signals and args for vorbis; add some docs for vorbis
+ * common:
+ Automatic update of common submodule
+ From 9cf8c9b to a6ce5c6
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * common/gstdoc-scangobj:
- * common/gtk-doc-plugins.mak:
- * docs/libs/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins.args:
- * docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- * docs/plugins/gst-plugins-base-plugins.signals:
- only scanobj stuff from our source module. Not sure yet
- if that's correct, given the hierarchy stuff :)
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
+ Fixes: #565777
-2005-09-15 Wim Taymans <wim@fluendo.com>
+2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- And enable 24 bits mode as well..
+ * tests/icles/stress-playbin.c:
+ stress-playbin: print the current uri
+ Print the current uri so that we can more easily see what uri caused a crash or
+ error.
-2005-09-15 Wim Taymans <wim@fluendo.com>
+2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_get_type), (gst_basertppayload_base_init),
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_finalize), (gst_basertppayload_setcaps),
- (gst_basertppayload_chain), (gst_basertppayload_set_options),
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
- (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Added rtp payloader base class.
+ * tests/icles/stress-playbin.c:
+ Print the errors more clearly
+ Print some more verbose messages when dealing with errors.
-2005-09-15 Andy Wingo <wingo@pobox.com>
+2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac (plugindir): Remove the EOL matcher from the
- regexp, as it causes me problems. Libtool? Make? Who knows?
+ * gst/playback/gstplaybin2.c:
+ Release the group lock when setting states
+ Release the group lock while we perform the state changes on the uridecodebins
+ because that might trigger callbacks that we need to handle with the group lock
+ taken. Avoids a possible deadly embrace in some id3/flac files.
+ Fixes #567396.
-2005-09-14 David Schleef <ds@schleef.org>
+2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/generic/states.c:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init):
- Fixes for changes in registry API.
+ * gst/playback/gstdecodebin2.c:
+ Combine finding and creating groups
+ Combine the search for the current group and optionally creating one into one
+ function so that we can avoid taking the lock multiple times.
- * configure.ac: Only export gst_plugins_desc. Add -no-undefined
- to GST_PLUGIN_LDFLAGS.
- * ext/libvisual/visual.c: Make the library shut up.
- * gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
- * gst-libs/gst/audio/gstaudiofilter.c: same
+2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ Playbin2: Don't leave unused parameters in debug statements.
+ Fixes build on macosx
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/inspect/plugin-libvisual.xml:
- * docs/plugins/tmpl/element-tcpserversink.sgml:
- * ext/theora/theoraenc.c:
- add libvisual plugin and theoraenc element to docs
+2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/theora/theoraenc.c:
- add theoraenc
+2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-13 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ Add some G_UNLIKELY because we can
+ Add a G_UNLIKELY when checking the shutdown variable.
- * gst/audioconvert/Makefile.am:
- Audioconvert derives from GstBaseTransform and should
- link to the library with our base elements to avoid
- unresolved symbols. Makes things work with MinGW (#316160)
+2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
- * gst/playback/test4.c: (main):
- Fix MinGW build problem and use g_usleep() instead of
- sleep() (#316162)
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ mixer interface: Add flags to enhance mixer interfaces
+ This patch adds a few flags to the mixer and mixerctrl interface to
+ better support OSSv4 (and potentially other backends).
+ Patch By: Garret D'Amore <garrett.damore@sun.com>
+ Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
+ API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
+ API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
+ API: GST_MIXER_TRACK_WHITELIST
-2005-09-12 Wim Taymans <wim@fluendo.com>
+2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/audioconvert/audioconvert.c: (float),
- (audio_convert_prepare_context), (audio_convert_convert):
- * gst/audioconvert/audioconvert.h:
- Cleanups, speedups, simplifications, added back support
- for 24 bits.
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: Fix strict aliasing error using a union
-2005-09-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/tmpl/element-tcpserversink.sgml:
- * gst/ffmpegcolorspace/gstffmpeg.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- add more elements to the docs
-
-2005-09-09 Jan Schmidt <thaytan@mad.scientist.com>
-
- * check/Makefile.am:
- * check/pipelines/simple_launch_lines.c: (setup_pipeline),
- (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
- Add extra tests for basetransform based components.
- Comment out the test_element_negotiation test until we decide
- if it's testing correct behaviour.
- * ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
- (gst_visual_chain), (gst_visual_change_state):
- Slightly more correct but still bogus timestamping.
- Fix state change function.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init):
- * gst/audioresample/gstaudioresample.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_class_init):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_prepare_size), (gst_videoscale_set_caps),
- (gst_videoscale_prepare_image):
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform_ip):
- Basetransform updates. Enable passthrough modes.
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
- (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
- Negotiation fix that allows the window to return to the original
- size and renegotiate passthrough upstream. Extra debug output.
-
-2005-09-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/sine/gstsinesrc.c:
- * gst/volume/gstvolume.c:
- fix up header include
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Fix a strict aliasing warning
+ Fix strict aliasing warnings from casting a sockaddr_storage and
+ using it as a sockaddr_in6. Use a union instead.
-2005-09-09 Stefan Kost <ensonic@users.sf.net>
+2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit):
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform):
- fixing lost sync, some more debugging
+ * docs/libs/.gitignore:
+ * docs/libs/tmpl/.gitignore:
+ * docs/plugins/.gitignore:
+ * docs/plugins/tmpl/.gitignore:
+ Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
-2005-09-08 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
- (gst_xvimagesink_check_xshm_calls):
- Fix compilation when XShm is not available.
+ * docs/plugins/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/gstvorbisenc.h:
+ * ext/vorbis/gstvorbisparse.h:
+ * ext/vorbis/gstvorbistag.h:
+ * ext/vorbis/vorbis.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisenc.h:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbisparse.h:
+ * ext/vorbis/vorbistag.c:
+ * ext/vorbis/vorbistag.h:
+ vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
-2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_dispose),
- (gst_visual_getcaps), (gst_visual_src_setcaps),
- (gst_visual_sink_setcaps), (get_buffer), (gst_visual_chain),
- (gst_visual_change_state):
- Finish fixing up libvisual plugin so that it runs.
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add conversion from/to YVYU colorspace
+ Fixes bug #572872.
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- gsttaginterface.h -> gsttagsetter.h
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
+ The conversion from UYVY to RGB24 and then to GRAY8
+ is quite slow. Fixes bug #569655.
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- added another test that failes for me (test is not active by default)
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix deadlock when shutting down. Fixes #572577.
-2005-09-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- v4l2 is no longer in gst-plugins-base
+ * tests/icles/stress-playbin.c:
+ stress-playbin: make more flexible, e.g. also useful for playbin2
-2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- In the output at the end, don't show the first plugin on the same
- line as "Core plug-ins, always built:".
- Indent the output as for other plugin categories
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- #define that can be used to not use peer buffer_alloc functions for
- test purposes.
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximage_buffer_get_type), (gst_ximagesink_ximage_new),
- (gst_ximagesink_show_frame):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
- (gst_xvimage_buffer_get_type), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_show_frame):
- Error case handling fixes. gst-launch fakesrc ! x[v]imagesink now
- fails gracefully instead of XError aborting or deadlocking.
-
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Match WSAStartup and WSACleanup correctly
+ Don't randomly call WSAStartup and WSACleanup but instead call the startup when
+ we create a connection and cleanup when we free it again. Because the internal
+ datastructure is refcounted, this should not cause any refcounting leaks when
+ the connection is managed correctly.
+ Fixes #562794.
- * ext/libvisual/Makefile.am:
- link against gst-base-libs
+2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-09-06 David Schleef <ds@schleef.org>
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
- * configure.ac: Enable libvisual plugin.
- * ext/libvisual/Makefile.am:
- * ext/libvisual/visual.c: Fixes to make it compile.
+2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
-=== release 0.9.2 ===
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-audio-uninstalled.pc.in:
+ * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
+ * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
+ * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
+ * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-riff-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-tag-uninstalled.pc.in:
+ * pkgconfig/gstreamer-video-uninstalled.pc.in:
+ Add srcdir to includes for out-of-source builds
+ When you use gstreamer uninstalled and build outside
+ the source tree, the includes need to be specified for
+ both the source tree and the build tree.
+ Signed-off-by: David Schleef <ds@schleef.org>
-2005-09-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
- * NEWS:
- * RELEASE:
* configure.ac:
- * docs/random/ChangeLog-0.8:
- releasing 0.9.2, "Spoon"
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Use shave for the build output
-2005-09-05 Michael Smith <msmith@fluendo.com>
+2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init):
- libvorbis accepts quality as low as -0.1, not just 0.0. So accept
- that in the vorbisenc element.
+ * win32/common/libgstrtsp.def:
+ win32: Add new symbol to libgstrtsp.def
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
- fix distcheck
- * gst/audioresample/resample.c:
- fix wrong docstring
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ * gst-libs/gst/rtsp/gstrtspextension.h:
+ Add method for handling server requests
+ Add a receive_request so that extensions can react to server requests.
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * common/gst-xmlinspect.py:
- * common/gtk-doc-plugins.mak:
- only inspect plugins for this given package
- require gst-python 0.9
+ * tests/check/libs/netbuffer.c:
+ Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
-2005-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Makefile.am:
- * autogen.sh:
- * common/gst-xmlinspect.py:
- * configure.ac:
- * docs/Makefile.am:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/tmpl/element-gnomevfssink.sgml:
- * docs/plugins/tmpl/element-multifdsink.sgml:
- * docs/plugins/tmpl/element-tcpserversink.sgml:
- * docs/plugins/tmpl/element-vorbisenc.sgml:
- * gst-plugins-base.spec.in:
- various doc-related updates
+ * ext/theora/theoraparse.c:
+ theoraparse: Use the correct unref functions
-2005-08-31 Wim Taymans <wim@fluendo.com>
+2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Resync if the buffer timestamps drift more than a 10th
- of a second.
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
+
+2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-08-31 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Unref the actual buffer instead of the memory address of the buffer
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- The 'timestamp-offset' property is registered as an int64, so
- let's use g_value_{set|get}_int64() in our setter and getter
- functions (makes it work and fixes warnings with gst-inspect).
+2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-30 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 5d7c9cc to 9cf8c9b
- * check/elements/audioconvert.c: (setup_audioconvert):
- * check/elements/audioresample.c: (setup_audioresample):
- * check/elements/volume.c: (setup_volume):
- Fix checks.
+2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgstvideo.def:
+ win32/common: Update .def files for recent API addition
- * common/gtk-doc-plugins.mak:
- * common/plugins.xsl:
- * docs/plugins/Makefile.am:
- make module a param
+2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/libs/rtp.c:
+ tests: Fix indentation
- * examples/seeking/seek.c: (make_mp3_pipeline),
- (make_mpeg_pipeline), (seek_cb), (start_seek), (stop_seek),
- (play_cb), (pause_cb), (stop_cb):
- update the example
+2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/video/video.c:
+ libs/video: Fix gst_video_format_new_caps* functions.
+ Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
+ don't add anything.
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform):
- do not update controlled params, if buffer has no timestamp
+2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ Automatic update of common submodule
+ From 80c627d to 5d7c9cc
- * configure.ac:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- controllerized elements also need to link against controller-libs ;)
+2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ Improve key/value parsing
+ Improve header field parsing by keeping a ref to the key/value instead of
+ copying it into a local variable.
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstgconf.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_create):
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform):
- controllerized two audio plugins
+2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-29 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Add trailing \0 to message length
+ We always put a trailing 0 at the end of the message body. Reflect this fact in
+ the length of the message.
- * ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push)
- (vorbis_handle_data_packet): Fix some int overflow errors.
+2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_init): Init total_time to
- -1.
- (gst_ogg_demux_perform_seek): Clamp segment_stop only if it's
- valid.
- (gst_ogg_pad_submit_packet): Subtract the chain's begin_time only
- if it's valid. Fixed streaming-mode playback.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Don't parse headers for data messages
+ Don't try to parse the headers on a data message because they don't have
+ headers.
- * check/elements/volume.c (cleanup_volume): Fix for running
- CK_FORK=no.
+2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
- * check/elements/audioconvert.c: Convert from native endian, not
- little endian.
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theoraenc: Add property for speed level control
+ Add property "speed-level" to control the amount of motion searching
+ the encoder does. This is only available in libtheora >= 1.0 and
+ will silently fail with earlier libraries. Fixes: #572275.
+ Signed-off-by: David Schleef <ds@schleef.org>
-2005-08-29 Michael Smith <msmith@fluendo.com>
+2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/Makefile.am:
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_get_type), (free_stream),
- (gst_ogg_parse_delete_all_streams), (gst_ogg_parse_new_stream),
- (gst_ogg_parse_find_stream), (gst_ogg_parse_base_init),
- (gst_ogg_parse_class_init), (gst_ogg_parse_init),
- (gst_ogg_parse_dispose), (gst_ogg_parse_submit_buffer),
- (gst_ogg_parse_append_header), (gst_ogg_parse_is_header),
- (gst_ogg_parse_buffer_from_page), (gst_ogg_parse_chain),
- (gst_ogg_parse_change_state), (gst_ogg_parse_plugin_init):
- Add an ogg parser element.
-
-2005-08-28 Andy Wingo <wingo@pobox.com>
-
- * Updates for two-arg init from GST_BOILERPLATE_FULL.
-
-2005-08-26 Wim Taymans <wim@fluendo.com>
-
- * gst/audioconvert/audioconvert.c: (if), (float),
- (audio_convert_get_func_index), (check_default),
- (audio_convert_clean_fmt), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_get_sizes),
- (audio_convert_convert):
- Cleanups.
-
-2005-08-26 Wim Taymans <wim@fluendo.com>
-
- * gst/audioconvert/audioconvert.c: (if), (float),
- (audio_convert_get_func_index), (check_default),
- (audio_convert_clean_fmt), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_get_sizes),
- (audio_convert_convert):
- More elegant and working temp buffer selection algo.
-
-2005-08-26 Wim Taymans <wim@fluendo.com>
-
- * gst/audioconvert/audioconvert.c: (if), (float),
- (audio_convert_get_func_index), (check_default),
- (audio_convert_clean_fmt), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_get_sizes),
- (get_temp_buffer), (audio_convert_convert):
- Use realloc else we lose our original data.
-
-2005-08-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Fix 'Since' tags
- * gst/audioresample/gstaudioresample.c:
- use base class' newsegment to properly timestamp
+2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-26 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add flags for interlaced video along with convenience methods for interlaced caps.
+ These three flags allow all know combinations of interlaced formats. They should
+ only be used when the caps contain 'interlaced=True'.
+ Fixes #163577 (yes, it's a 4 year old bug).
- * gst/audioconvert/audioconvert.c: (if), (float),
- (audio_convert_get_func_index), (check_default),
- (audio_convert_clean_fmt), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_get_sizes),
- (get_temp_buffer), (audio_convert_convert):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
- (gst_audio_convert_transform_caps),
- (gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
- Oops, allocate enough space to perform the channel mix.
+2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-26 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Make RTSPConnection opaque and rename RTSPChannel
+ Make the RTSPConnection object opaque so that we can extend it in the future.
+ Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/audioconvert.c: (if), (float),
- (audio_convert_get_func_index), (check_default),
- (audio_convert_clean_fmt), (audio_convert_prepare_context),
- (audio_convert_clean_context), (audio_convert_get_sizes),
- (get_temp_buffer), (audio_convert_convert):
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_init),
- (gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
- (gst_audio_convert_get_unit_size),
- (gst_audio_convert_transform_caps),
- (gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
- (gst_audio_convert_transform_ip), (gst_audio_convert_transform):
- * gst/audioconvert/gstaudioconvert.h:
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
- (gst_channel_mix_fill_identical),
- (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
- (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
- (gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
- (gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
- (gst_channel_mix_mix):
- * gst/audioconvert/gstchannelmix.h:
- Cleanups, librarify a bit, optimize, better negotiation and more.
+2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/riff/riff-media.c:
+ Add some more mappings for h264 in riff
+
+2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (ogg_find_peek):
- Another from MikeS:
- During typefinding, don't support negative offsets
- (offsets from the end of the stream) in our typefind->peek() function
- - nothing embedded in ogg ever needs them. However, we need to recognise
- those requests and reject them, otherwise we return invalid pointers.
+ * win32/common/libgstrtsp.def:
+ Add new RTSP symbols to def files
+ Add the new RTSP symbols to the windows def file.
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
- (vorbisdec_finalize), (vorbis_handle_type_packet):
- Big shout-out to MikeS for fixing this giant memory leak.
- Huzzah!
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/appsink.c:
+ Add method to install callbacks on appsink
+ Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
+ Fixes #571299.
+ Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
+ performant alternative to connecting to the signals.
+ Add a unit test for appsink.
+ Clean up some of the appsink docs.
+ API: GstAppSink::gst_app_sink_set_callbacks()
+
+2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Add RTSP accept method
+ Add a method to accept a connection on a socket and create a GstRTSPConnection
+ for it.
+ API: gst_rtsp_connection_accept()
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose),
- (audio_convert_get_unit_size):
- plug some leaks
+2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Add RTSP channel object for async io
+ Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
+ that the connection can be monitored from a maincontext. This allows us to
+ operate in ASYNC mode, which is handy when building a server.
+ Rework the old code to use the async code under the hood.
+ API: gst_rtsp_channel_new()
+ API: gst_rtsp_channel_unref()
+ API: gst_rtsp_channel_attach()
+ API: gst_rtsp_channel_queue_message()
+
+2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * check/Makefile.am:
- * check/elements/audioconvert.c: (setup_audioconvert),
- (cleanup_audioconvert), (get_int_caps), (verify_convert),
- (GST_START_TEST), (audioconvert_suite), (main):
- add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
- note that for buffers of 1/3 sec this means DURATION(c) is
- one nanosecond more than for a and b
-
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check/Makefile.am:
- * check/elements/audioresample.c: (setup_audioresample),
- (cleanup_audioresample), (fail_unless_perfect_stream),
- (test_perfect_stream_instance), (GST_START_TEST),
- add a check for audioresample
- (audioresample_suite), (main):
- * check/elements/volume.c: (GST_START_TEST):
- remove unused method
- * gst/audioresample/gstaudioresample.c:
- set correct buffer parameters since we're changing them
- * gst/audioresample/resample_ref.c: (resample_scale_ref):
- add some debug
+ audioresample: Add locking to protect the resampling context
+ When setting the quality/filter-length while PLAYING the
+ resampling context will be destroyed and created again in
+ some cases, which will cause crashes in the transform function
+ if it's called at that time.
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioresample/debug.c:
- * gst/audioresample/gstaudioresample.c:
- add room for extra overlap samples when asked to transform size
- protect against possible mem corruption and check for discrepancies
- between written size and outbuffer's size so we can warn for
- potential problems
- * gst/audioresample/resample.c: (resample_init),
- (resample_get_output_size_for_input), (resample_get_output_size),
- (resample_set_n_channels), (resample_set_format):
- set debug level based on RESAMPLE_DEBUG env var
- make sure that get_output_size* returns a whole number of
- sample_size
- set sample_size each time either channel or format is set
- * gst/audioresample/resample_chunk.c: (resample_scale_chunk):
- * gst/audioresample/resample_functable.c:
- (resample_scale_functable):
- * gst/audioresample/resample_ref.c: (resample_scale_ref):
- remove r->sample_size, it's done in resample.c now
- add some debugging to the ref implementation
- make sure we only give back bytes that are wholes of the sample
- size
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/videotestsrc/videotestsrc.c:
+ ffmpegcolorspace/videotestsrc: Use v308 instead of V308
-2005-08-25 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- Revert unpopular change for GST_MESSAGE_SRC to GObject.
+2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-08-25 Stefan Kost <ensonic@users.sf.net>
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
+ Only conversions from/to are implemented, which
+ gives (indirect) support for all possible conversions.
+ Partially fixes bug #571147.
- * gst/volume/gstvolume.c:
- made set_caps function static
+2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
+ Partially fixes bug #571147.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_change_state):
- Stop leaking taglists.
+2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: don't abort when downstream pulls a buffer of size 0
+ Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
+ aborting. Fixes #571009 (wma file with ID3v2 tag).
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
- (gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
- (gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
- Parse seeking events better.
- Unref static caps.
- Generate correct newsegment events, fixes seeking in live oggs.
+2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_src_getcaps),
- (theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
- Use newsegment values to report correct play time.
+ * gst-libs/gst/riff/riff-read.c:
+ riff: error out on nonsensical chunk sizes instead of aborting
+ When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
+ continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
+ in g_malloc() or crash.
+ Fixes #553295, crash with fuzzed AVI file.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_sink_event):
- * ext/vorbis/vorbisdec.h:
- Parse and use newsegment values to report correct play time.
+2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- Clear ringbuffer on flush.
- Use newsegment values to calculate playback time.
+ * .gitignore:
+ Make git ignore backup files.
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
- Basesink does newsegment calculations for us now.
+2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ Revert "Remove pad-removed handlers after setting the decodebins to NULL."
+ This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
+ This brought back some deadlocks. A small leak is better, for now. Need to
+ figure out a way to fix the leak properly.
- * check/Makefile.am:
- * configure.ac:
- add core's plugins to the mix so that playbin works
- * check/generic/states.c: (GST_START_TEST):
- set a 0 timeout on pipelines, so they don't force the next
- state change
- * gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
- (gst_play_base_bin_change_state):
- remove the crappy error handling and do GST error handling
+2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix segfault on notify after group change.
+ If our group has been switched, then we get a selector active-pad
+ notification, we don't need to notify.
- * check/Makefile.am:
- * check/generic/states.c: (GST_START_TEST), (states_suite), (main):
- add same test as to core, it bitches out on playbin atm.
+2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playbin2: Look for volume/mute properties recursively in audio element.
+ Rather than only checking for volume property on the audio sink
+ directly, recursively look for it on sinks within it (if it's a bin).
+ Allows use of sink-as-volume-control where the application has supplied
+ an audio-sink bin that includes a real audio sink internally.
- * configure.ac:
- Remove audioscale.
+2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.spec.in:
+ Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
- (gst_videoscale_prepare_size), (parse_caps),
- (gst_videoscale_set_caps), (gst_videoscale_get_size),
- (gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
- (gst_videoscale_transform):
- * gst/videoscale/gstvideoscale.h:
- Refactor, make use of BaseTranform really well.
+2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
+ Partially fixes bug #571147.
- * configure.ac:
- compile audioresample
- * gst/audioresample/Makefile.am:
- * gst/audioresample/buffer.c:
- * gst/audioresample/functable.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- * gst/audioresample/resample.c:
- (resample_get_output_size_for_input):
- * gst/audioresample/resample.h:
- * gst/audioresample/resample_chunk.c:
- * gst/audioresample/resample_functable.c:
- * gst/audioresample/resample_ref.c:
- port to use basetransform; doesn't work in all cases yet
+2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ gstrtspmessage: Minor documentation correction.
+ Corrected documentation about what needs to be freed after calling
+ gst_rtsp_message_new(), gst_rtsp_message_new_request(),
+ gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_init),
- (audio_convert_get_unit_size), (audio_convert_transform_caps),
- (audio_convert_fixate_caps), (audio_convert_set_caps),
- (audio_convert_transform),
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format),
- (gst_audio_convert_channels):
- * gst/audioconvert/gstchannelmix.c:
- * gst/audioconvert/gstchannelmix.h:
- port to basetransform
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_get_unit_size):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_transform_caps), (gst_videoscale_get_unit_size):
- fix for basetransform changes
+2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
-2005-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: Fix race condition that made alsamixer not working properly
+ This is due to race conditions between functions that
+ modified the mixer like set_volume and
+ snd_mixer_handle_events since the handle_events
+ can now be called at any time.
+ Fixed by adding locking around any snd_mixer call
+ since even read functions can modify the mixer stucture, since
+ alsa likes to clear it's values before reading new ones.
+ The favorite race condition seemed to be that set_volume
+ called read_elem (in alsalib) that reset the volumes to
+ 0 and then read them with read_x_volume. This read looped
+ on each channel and as the race condition occured the
+ channels value could be anything , most of the time
+ it was 0. Thus no value was read or only the value of
+ one channel was and the volume was reset to 0.
+ Fixes bug #478512.
+
+2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Bump revision to use for common submodule.
+
+2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
- * check/Makefile.am:
- Add CHECK_CFLAGS and LDFLAGS
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: do not call _xwindow_clear on ready->paused.
+ Calling clear at that transition does things like stopping xvideo (which is not
+ running at that time) and also clearing anything what the application might have drawn.
+ This breaks handle-expose and autopaint-colorkey features.
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- GST_MESSAGE_SRC became a GObject
+2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ * gst-libs/gst/rtsp/gstrtsprange.h:
+ RTSPRange: Add method to serialize ranges
+ Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
+ be used by a server.
+ API: GstRTSPRange::gst_rtsp_range_to_string()
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
- (gst_ring_buffer_clear_all):
- * gst-libs/gst/audio/gstringbuffer.h:
- Added function to clear the ringbuffer.
+2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-24 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ GstRTSPUrl: Add some const to methods
+ Add const to the methods that do not modify the object.
- * sys/v4l/gstv4lelement.c (gst_v4lelement_start)
- (gst_v4lelement_stop): Call _start and _stop for xoverlay instead
- of _open and _close.
+2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
+ The flags where present but actually not been taken into account.
+
+2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/gstv4lxoverlay.c (gst_v4l_xoverlay_set_xwindow_id): Open
- an Xv connection here, instead of all the time. Make Xv only be
- loaded if you axe for it. Kindof a workaround for buggy behaviour
- of Xv when using remote xservers (XvQueryExtension would block).
- (gst_v4l_xoverlay_stop, gst_v4l_xoverlay_start): New functions,
- replace the _open and _close public API. Only start the xv
- connection if necessary.
- (gst_v4l_xoverlay_open, gst_v4l_xoverlay_close): Made static.
-
-2005-08-23 David Schleef <ds@schleef.org>
-
- * gst/audioresample/Makefile.am: Leet audioresampling code
- * gst/audioresample/buffer.c:
- * gst/audioresample/buffer.h:
- * gst/audioresample/debug.c:
- * gst/audioresample/debug.h:
- * gst/audioresample/functable.c:
- * gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- * gst/audioresample/resample.c:
- * gst/audioresample/resample.h:
- * gst/audioresample/resample_chunk.c:
- * gst/audioresample/resample_functable.c:
- * gst/audioresample/resample_ref.c:
+ audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
+ The comment will ensure that is is marked properly in the docs and the
+ GParamSpecflag was causing a duplicated initialisation of the same value.
-2005-08-23 Wim Taymans <wim@fluendo.com>
+2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * examples/seeking/seek.c: (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
- Small seek updates.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Add more g_return_if_fail() calls
+ Check that we have a valid file descriptor before entering certain functions in
+ order to avoid undesirable situations.
+ Add some more debugging in the connect method.
-2005-08-23 Andy Wingo <wingo@pobox.com>
+2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Only fixate endianness if it is
- present in the caps.
+ * configure.ac:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Only pull in liboil if its actualy used.
+ Liboil still has quite significant startup overhead especialy on embedded
+ platforms. In audioresample it was only used for the profiling timer.
-2005-08-22 Andy Wingo <wingo@pobox.com>
+2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsasink.c (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
- device-name property.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
+ Add comments about the flac format. Tighten the check to not allow values that
+ refer to headers.
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
- close_device in the ring buffer, like gstaudiosink.
+2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
- macro to implement the interface without much code. Cleanups.
+ * win32/common/libgstrtsp.def:
+ Add new methods
+ Add new methods to the windows def file.
- * ext/alsa/gstalsasrc.h:
- * ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
- READY.
-
- * ext/alsa/Makefile.am: Add new files.
- * ext/alsa/gstalsamixerelement.c:
- * ext/alsa/gstalsamixerelement.c: Split element code out from
- mixer code so that alsasrc can be a mixer too.
-
-2005-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check/elements/volume.c: (setup_volume), (cleanup_volume),
- (GST_START_TEST):
- * check/elements/vorbisdec.c: (setup_vorbisdec),
- (cleanup_vorbisdec), (GST_START_TEST), (vorbisdec_suite):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_handle_identification_packet),
- (vorbis_handle_comment_packet), (vorbis_handle_type_packet),
- (vorbis_handle_header_packet), (vorbis_dec_push),
- (vorbis_dec_chain):
- use the setup/teardown methods to save code. save code is good.
-
-2005-08-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check/Makefile.am:
- add ext dir for plugins
- add vorbisdec test conditionally
- * check/elements/volume.c: (setup_volume), (cleanup_volume),
- (GST_START_TEST), (volume_suite):
- add a test with wrong caps
- * check/elements/vorbisdec.c: (chain_func), (setup_vorbisdec),
- (cleanup_vorbisdec), (GST_START_TEST), (vorbisdec_suite), (main):
- add a vorbisdec test
- * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream),
- (gst_ogg_demux_chain), (gst_ogg_demux_loop):
- clean up debug output
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- yay, fix a segfault/security issue in vorbisdec
- gst-launch fakesrc ! vorbisdec wasn't happy
- * ext/vorbis/vorbisenc.c: (vorbisenc_get_type),
- (gst_vorbisenc_class_init), (gst_vorbisenc_sink_setcaps),
- (gst_vorbisenc_convert_src), (gst_vorbisenc_convert_sink),
- (gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
- (gst_vorbisenc_init), (gst_vorbisenc_metadata_set1),
- (gst_vorbisenc_set_metadata), (get_constraints_string),
- (update_start_message), (gst_vorbisenc_setup),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
- (gst_vorbisenc_push_packet), (gst_vorbisenc_sink_event),
- (gst_vorbisenc_chain), (gst_vorbisenc_get_property),
- (gst_vorbisenc_set_property), (gst_vorbisenc_change_state):
- * ext/vorbis/vorbisenc.h:
- march in line
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- have the kow come home
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
- debug my func ptr
- * gst/volume/gstvolume.c: (volume_set_caps):
- add a debug
+2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-08-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: remove duplicate detail strings when calling the external codec installer
+ It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
- * Makefile.am:
- * check/.cvsignore:
- * check/Makefile.am:
- * check/elements/.cvsignore:
- * check/elements/volume.c: (chain_func), (event_func),
- (setup_volume), (cleanup_volume), (GST_START_TEST), (volume_suite),
- (main):
- * configure.ac:
- add unit test structure for gst-plugins-base
- add a test for volume
- * gst/volume/gstvolume.c: (gst_volume_list_tracks),
- (gst_volume_set_volume), (gst_volume_get_volume),
- (gst_volume_set_mute), (gst_volume_class_init), (gst_volume_init),
- (volume_funcfind), (volume_process_float), (volume_process_int16),
- (volume_set_caps), (volume_transform), (volume_update_mute),
- (volume_update_volume), (volume_set_property),
- (volume_get_property):
- document a little; use basetransform vmethod _set_caps
-
-2005-08-19 Andy Wingo <wingo@pobox.com>
+2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsamixertrack.h:
- * ext/alsa/gstalsamixertrack.c:
- * ext/alsa/gstalsamixeroptions.h:
- * ext/alsa/gstalsamixeroptions.c:
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixer.c: Port to 0.9.
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
- * ext/alsa/Makefile.am: Build mixer, mixeroptions, mixertracks.
- Remove gstalsa.c and alsaclock. No more cruft here.
-
-2005-08-18 Wim Taymans <wim@fluendo.com>
+2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain),
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (gst_base_rtp_depayload_queue_release):
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Fix for RTPBuffer changes.
-
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data),
- (gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data),
- (gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len),
- (gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len),
- (gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data),
- (gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len),
- (gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version),
- (gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding),
- (gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to),
- (gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension),
- (gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc),
- (gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc),
- (gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker),
- (gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type),
- (gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq),
- (gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp),
- (gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len),
- (gst_rtpbuffer_get_payload):
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Don't subclass GstBuffer but add methods and helper functions
- to construct and manipulate RTP packets in regular GstBuffers.
+ * configure.ac:
+ * gst/audioresample/gstaudioresample.c:
+ Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
-2005-08-18 Stefan Kost <ensonic@users.sf.net>
+2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
- moved statement below switch
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- added debug ptr
+ * sys/ximage/ximagesink.c:
+ Fix buffer_alloc in ximagesink
+ Remove some useless debug info that reported wrong image sizes.
+ When upstream does not accept out suggested size, fall back to allocating an
+ image of the requested width/height instead of the currently configured size.
+ The problem is that an image is reused from the pool because the width/height
+ match but the caps on the new buffer are the requested caps with possibly
+ different height/width resulting in errors.
-2005-08-16 Wim Taymans <wim@fluendo.com>
+2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Open and close device in READY<->NULL state change.
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ Fix documentation for autoplug-select
+ fix the documentation strings for the autoplug-select signal.
+ Fixes #570142.
-2005-08-16 Andy Wingo <wingo@pobox.com>
+2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * examples/seeking/Makefile.am: Don't compile non-compiling
- compiled objects with the compiler.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ Fix string leak in rtspmessage
+ when we remove a header field from a message we must free the value associated
+ with the key to avoid a memory leak.
- * examples/seeking/seek.c (make_dv_pipeline): Update for new DV
- elements.
+2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-12 Philippe Khalaf <burger@speedy.org>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Made a thread to release the queue.
- Removed timestamp conversion for now.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ Its "Base Library" and not just "Library".
-2005-08-10 Philippe Khalaf <burger@speedy.org>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- Added rtp timestamp -> gst timestamp conversion.
- Fixed several problems with queue.
+2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ Link to the class, as we can't link to the members yet.
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/net/gstnetbuffer.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Add padding (you will need to rebuild gst-plugins-base,
- gst-plugins and all applications afterwards!)
+2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ Remove pad-removed handlers after setting the decodebins to NULL.
+ They do needed cleanup; without this we leak selector requestpads.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
- (gst_riff_parse_chunk):
- Fix bug in debug message and add some more debug messages.
+2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-08 Edward Hervey <edward@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Unref selector request pad even if we no longer have a selector.
+ During destruction, we won't have a selector any more, but we still need
+ to unref the pad to avoid leaking it.
- * gst-libs/gst/riff/riff-media.c:
- backported updates since branch
+2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-08 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstplaybin2.c:
+ Unref source in playbin2's finalize method
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_change_state): Open the device in NULL->READY
- like good elements should. Close on READY->NULL too.
+2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst-libs/gst/audio/gstaudiosink.c
- (gst_audioringbuffer_open_device,
- (gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
- (gst_audioringbuffer_release): Updates for new ring buffer API,
- hook into the new audio sink api.
+ * gst/playback/gstplaysink.c:
+ Fix more leaks of pads and elements in gstplaysink.
+ Don't keep extra references to volume and mute elements; we don't need
+ to do so.
+ Ensure we unref pads that we have references to, and release request
+ pads.
- * gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
- (GstAudioSinkClass.close): Just open and close the device -- no
- resource allocation or configuration.
- (GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
- vmethods, handle device setup and resource allocation.
+2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
- * ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
- (gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
- base class API.
+ * gst/playback/gstplaysink.c:
+ Avoid leaking all playsinks. Fix some internal leaks.
+ Playsink was holding references to itself. Don't do that, it's not cool.
+ Also, free all chains in dispose.
- * gst-libs/gst/audio/gstringbuffer.h
- (GstRingBufferClass.open_device, GstRingBufferClass.close_device):
- New vmethods.
+2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
- (gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
- New API functions. The device should be opened before acquiring
- and closed after releasing.
+ * gst/playback/gstplaybin2.c:
+ Unref peer request pad after releasing it, since we hold a reference.
-2005-08-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst-libs/gst/interfaces/mixer.h:
- Reset padding to GST_PADDING.
+ * gst/playback/gstplaybin2.c:
+ Fix caps leak in playbin2.
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst/playback/gstplaybin.c: (remove_sinks):
- Remove visualization from parent explicitely; works around some
- apparent refcount issue that I haven't tracked down yet.
+ * gst/playback/gstplaybin2.c:
+ Unref active pad from selector when finding active stream.
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
- * ext/alsa/gstalsasink.c: (set_hwparams):
- Assign debug category, add negotiation debug msgs.
+ * gst/playback/gstplaybin2.c:
+ Free uris when finalizing playbin2 instance.
-2005-08-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_start):
- Fix error code for file-not-found to NOT_FOUND.
+ * gst/playback/gsturidecodebin.c:
+ Unref pads when iterating over them in analyse_source.
+ Fixes leak of source's srcpad when using uridecodebin.
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- renamed to actual element names, so much nicer to look at
- * docs/plugins/tmpl/gstmultifdsink.sgml:
- remove
- * docs/plugins/tmpl/multifdsink.sgml:
- * docs/plugins/tmpl/tcpserversink.sgml:
- add
- * ext/alsa/gstalsa.c:
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
- * ext/ogg/gstoggmux.c:
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
- * gst/playback/gstdecodebin.c:
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
- * gst/tcp/gsttcpserversink.c:
- various fixes and documentation additions
+ Add releaseinfo with online url.
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
- * common/Makefile.am:
- * common/gstdoc-scangobj:
- * common/gtk-doc-plugins.mak:
- * common/gtk-doc.mak:
- add a custom scangobj that uses the registry
- add a custom gtk-doc-plugins.mak that uses it
- some doc build fixes
- * configure.ac:
- * docs/Makefile.am:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.types:
- * docs/plugins/tmpl/gstmultifdsink.sgml:
- add docs for one element, multifdsink
- * gst/adder/gstadder.h:
- * gst/volume/gstvolume.h:
- don't privatize enum
- * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_sync_method_get_type), (gst_client_status_get_type),
- (gst_multifdsink_class_init),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
- (gst_tcpclientsink_init), (gst_tcpclientsink_setcaps),
- (gst_tcpclientsink_render):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
- (gst_tcpclientsrc_init), (gst_tcpclientsrc_create),
- (gst_tcpclientsrc_start):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
- (gst_tcpserversrc_init), (gst_tcpserversrc_create):
- * gst/tcp/gsttcpserversrc.h:
- * gst/typefind/gsttypefindfunctions.c:
- remove superfluous Type stuff
+ * gst/playback/gstplaybasebin.c:
+ Fix compilation warning on Forte
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
- * gst/playback/gstplaybin.c: (gen_video_element):
- Enable videoscale.
+ * gst/adder/gstadder.c:
+ Don't do void pointer arithmetic.
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/gconf/gconf.c:
- * gst-libs/gst/gconf/gconf.h:
- Fix some Andy Problem [tm].
+ * common:
+ Bump common
-2005-08-04 Andy Wingo <wingo@pobox.com>
+2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c
- (gst_ffmpegcsp_get_size): Adapt to API changes.
+ * autogen.sh:
+ * common:
+ Use a symbolic link for the pre-commit client-side hook
- * gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
- Implement an in-place do-nothing transform.
+2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * .gitignore:
+ Add more files/directories to ignore
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
- (gst_ximagesink_renegotiate_size):
- Do not set new window sizes yet if we prepare a new buffer size
- for upstream renegotiation (software scaling) at some point in the
- future, because this new size waqs not actually accepted yet. Once
- accepted, renegotiation later on will set the new sizes just fine.
- Fixes a videotestsrc ! queue ! videoscale ! ximagesink xoverlay
- embedding testcase.
+2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-03 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ fix some typos
+ Fix some typos in the doc string of the new
+ gst_rtsp_options_as_string() method.
- * sys/ximage/ximagesink.c (gst_ximagesink_renegotiate_size):
- (gst_ximagesink_buffer_alloc):
- Protect the height, width, and desired_caps with the pool_lock.
- Fixes videotestsrc ! queue ! ximagesink.
+2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-02 Edward Hervey <edward@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ Add new RTSP message method to set header
+ Add gst_rtsp_message_take_header() that takes ownership of the passed header
+ value. This allows us to avoid an allocations and memory copy in some
+ situations.
+ API: GstRTSPMessage::gst_rtsp_message_take_header()
- * gst/volume/gstvolume.c:
- include left from controller cleanup
+2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-02 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_change_state):
- Stop collectpads before calling the parent state
- change function on PAUSED->READY.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add new method to docs
+ Add the new gst_rtsp_options_as_text() method to the docs.
-2005-08-01 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- When testing for X libs, use the X CFlags
- * gst/adder/gstadder.c: (gst_adder_change_state):
- Stop the collectpads before calling parent state change function
- on PAUSED->READY, otherwise we deadlock deactivating pads.
+2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-01 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ Add method to serialize RTSP options
+ Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
+ string.
+ API: GstRTSP::gst_rtsp_options_as_text()
- * configure.ac:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * examples/Makefile.am:
- * gst/sine/Makefile.am:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_init), (gst_sinesrc_create),
- (gst_sinesrc_set_property), (plugin_init):
- * gst/sine/gstsinesrc.h:
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (gst_volume_set_volume),
- (gst_volume_set_mute), (gst_volume_dispose), (gst_volume_init),
- (volume_process_float), (volume_process_int16),
- (volume_set_property), (plugin_init):
- * gst/volume/gstvolume.h:
- deactivate and remove dparams (libgstcontrol)
+2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-07-29 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ Ensure we have sufficient data when using data scan contexts.
+ Fixes crashes typefinding things that look like they might contain AAC
+ data (but probably aren't actually AAC).
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
- Convert me to BaseTransform!! help..
+2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-07-29 Andy Wingo <wingo@pobox.com>
+ * ext/gio/Makefile.am:
+ Fix include order for gio plugin
- * ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
- sinks.
+2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
- * ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
- support of both endiannesses.
+ * win32/common/config.h:
+ Update win32 config.h for 0.10.22.1 dev cycle
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix confusing debug message (s/event/query/)
+ * .gitignore:
+ * docs/libs/.gitignore:
+ * gst-libs/gst/audio/.gitignore:
+ * gst-libs/gst/video/.gitignore:
+ * po/.gitignore:
+ * tests/examples/dynamic/.gitignore:
+ Extend and clean up git ignores
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.h:
- Use "_stdint.h" instead of <stdint.h>
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/README:
+ * gst/audioresample/arch.h:
+ * gst/audioresample/buffer.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/debug.c:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/fixed_arm4.h:
+ * gst/audioresample/fixed_arm5e.h:
+ * gst/audioresample/fixed_bfin.h:
+ * gst/audioresample/fixed_debug.h:
+ * gst/audioresample/fixed_generic.h:
+ * gst/audioresample/functable.c:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample.h:
+ * gst/audioresample/resample_chunk.c:
+ * gst/audioresample/resample_functable.c:
+ * gst/audioresample/resample_ref.c:
+ * gst/audioresample/resample_sse.h:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_double.c:
+ * gst/audioresample/speex_resampler_float.c:
+ * gst/audioresample/speex_resampler_int.c:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ * gst/speexresample/Makefile.am:
+ * gst/speexresample/README:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/fixed_arm4.h:
+ * gst/speexresample/fixed_arm5e.h:
+ * gst/speexresample/fixed_bfin.h:
+ * gst/speexresample/fixed_debug.h:
+ * gst/speexresample/fixed_generic.h:
+ * gst/speexresample/gstspeexresample.c:
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/resample.c:
+ * gst/speexresample/resample_sse.h:
+ * gst/speexresample/speex_resampler.h:
+ * gst/speexresample/speex_resampler_double.c:
+ * gst/speexresample/speex_resampler_float.c:
+ * gst/speexresample/speex_resampler_int.c:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/audioresample.c:
+ * tests/check/elements/speexresample.c:
+ Rename files and types from speexresample to audioresample
+ Rename files and types from speexresample to audioresample
+ to finish the move and to prevent any confusion.
-2005-07-27 Wim Taymans <wim@fluendo.com>
+2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/Makefile.am:
- Revert wrong commit.
+ * sys/xvimage/xvimagesink.c:
+ Add some more debugging to the Xv strides
+ Add some more debugging to the strides as they are received from the server and
+ the expected strides.
-2005-07-27 Wim Taymans <wim@fluendo.com>
+2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
- More compilation fixen.
+ * gst/typefind/gsttypefindfunctions.c:
+ Add typefind function for gsm
+ Because core now supports typefindfactories without a typefind function we can
+ register a factory fo GSM that will --if all else fails-- assume the file is a
+ GSM file based on the registered extension.
+ Fixes #566661.
-2005-07-27 Wim Taymans <wim@fluendo.com>
+2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render),
- (gst_base_audio_sink_create_ringbuffer),
- (gst_base_audio_sink_change_state):
- Fix compilation.
-
-2005-07-27 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (setup_dynamic_link),
- (make_dv_pipeline), (make_vorbis_theora_pipeline), (query_rates),
- (query_positions_elems), (query_positions_pads), (do_seek):
- Update seek example.
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
- (gst_ogg_pad_typefind), (gst_ogg_demux_chain_elem_pad),
- (gst_ogg_demux_queue_data), (gst_ogg_demux_chain_peer),
- (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
- (gst_ogg_demux_handle_event),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_chain), (gst_ogg_demux_send_event),
- (gst_ogg_demux_loop):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
- * ext/theora/theoradec.c: (theora_dec_src_event),
- (theora_dec_src_getcaps), (theora_dec_sink_event),
- (theora_dec_push), (theora_dec_chain):
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_dec_push),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_event),
- (gst_vorbisenc_chain):
- * gst/playback/gststreaminfo.c: (cb_probe):
- * gst/subparse/gstsubparse.c: (gst_subparse_src_event):
- * gst/videorate/gstvideorate.c: (gst_videorate_event):
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_event):
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame),
- (gst_ximagesink_navigation_send_event):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Various event updates and cleanups
+ * gst/playback/gsturidecodebin.c:
+ Use more performant link function
+ We can use gst_element_link_pads() instead of the more generic
+ gst_element_link() function because we know the pads. This saves some cycles
+ because the more generic function needs to search for possible compatible caps
+ etc.
-2005-07-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_prepare_images):
- Fix segfault for I420/YV12.
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ Add more codec ids for RIFF formats
+ Handle codec ID for various other AAC formats.
+ Sync the list of possible codec ids with that of ffmpeg.
+ Fixes #567255
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
- Report bitrate.
+ * ext/theora/theoradec.c:
+ Use rounded values for image strides and sizes
+ Round up the height before calculating the expected size and
+ strides of the output image.
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- Switch to auto*sink elements as default sinks; add volume element
- so that volume control in totem works.
+ * ext/alsa/gstalsasink.c:
+ Improve debug message
+ Improve the debug message when alsa returns an error.
-2005-07-21 Wim Taymans <wim@fluendo.com>
+2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gen_preroll_element):
- * gst/playback/gstplaybin.c: (setup_sinks),
- (gst_play_bin_change_state):
- Refcount fix and more comments.
+ * gst-libs/gst/app/gstappsrc.c:
+ Reset queued_bytes counter when flushing
+ Set the amount of queued bytes in the internal queue back to 0 when we clear the
+ queue.
+ Fixes #567982
-2005-07-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
- * sys/ximage/Makefile.am:
- * sys/ximage/ximage.c: (plugin_init):
- * sys/ximage/ximagesink.c:
- Prepare for adding ximagesrc, rename of plugin to ximage etc.
-
-
-2005-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
- (gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
- (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_pad_submit_page), (gst_ogg_chain_new),
- (gst_ogg_demux_init), (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info),
- (gst_ogg_demux_collect_info), (gst_ogg_demux_chain),
- (gst_ogg_demux_send_event), (gst_ogg_demux_loop):
- Generate correct disconts for live chained oggs.
+ * gst/typefind/gsttypefindfunctions.c:
+ Add typefinder for Mobile XMF. Fixes bug #568707.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render),
- (gst_base_audio_sink_create_ringbuffer),
- (gst_base_audio_sink_change_state):
- Handle discont math correctly.
-
- * gst/playback/gstplaybin.c: (add_sink):
- Some small debug cleanup.
-
-2005-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_init), (gst_ogg_pad_event),
- (gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
- (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_pad_submit_page), (gst_ogg_chain_new),
- (gst_ogg_demux_init), (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_find_pad),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
- (gst_ogg_demux_send_event), (gst_ogg_demux_loop),
- (gst_ogg_demux_change_state), (gst_ogg_print):
- Reorganize code to send the right disconts when in streaming
- mode.
-
-2005-07-20 Andy Wingo <wingo@pobox.com>
-
- * gst/videoscale/vs_image.c (vs_image_scale_nearest_YUYV): Typo
- fix (?), fixes a seggie mcfalterson (#310894).
-
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_headers),
- (gst_ogg_mux_set_header_on_caps):
- * ext/theora/theoraenc.c: (theora_set_header_on_caps):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
- * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_set_channel_positions),
- (gst_audio_set_structure_channel_positions_list):
- * gst/playback/gstdecodebin.c: (dynamic_create):
- * gst/playback/gstplaybasebin.c: (setup_source), (mute_group_type):
- * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
- Fixes for API changes in core.
+2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ Fix linking on Solaris. Fixes bug #568482.
+ Check for nsl and socket libraries and add them to
+ LIBS if they're found. They're needed for socket()
+ and gethostbyname() on Solaris.
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- Use _new_custom() so we can set custom message types for buffering
- messages.
+2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/playback/gstplaybasebin.c:
+ Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/gconf/.cvsignore:
- * gst-libs/gst/gconf/Makefile.am:
- * gst-libs/gst/gconf/test-gconf.c:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-gconf-uninstalled.pc.in:
- * pkgconfig/gstreamer-gconf.pc.in:
- Remove gconf stuff, use gconf elements instead from now on.
+2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-07-20 Wim Taymans <wim@fluendo.com>
+ * common:
+ Update common snapshot.
- * gst-libs/gst/audio/TODO:
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
- (gst_audio_clock_get_internal_time):
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
- (gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
- (gst_base_audio_sink_render),
- (gst_base_audio_sink_create_ringbuffer),
- (gst_base_audio_sink_change_state):
- Make sure the audio clock always returns an increasing value.
+2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-19 Andy Wingo <wingo@pobox.com>
+ * common:
+ Fix pre-commit hook
- * gst/videotestsrc/: Cleanups.
+2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-07-19 Wim Taymans <wim@fluendo.com>
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Better debugging.
+2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-19 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/fft/gstfftf32.c:
+ * gst-libs/gst/fft/gstfftf64.c:
+ * gst-libs/gst/fft/gstffts16.c:
+ * gst-libs/gst/fft/gstffts32.c:
+ Reduce the number of allocations for creating FFT contexts
+ Reduce the number of allocations from 2 to 1 for every FFT
+ context by allocating enough memory for the FFT context
+ and passing parts of it to the kissfft allocation functions.
- * examples/seeking/seek.c: (make_dv_pipeline),
- (make_vorbis_theora_pipeline), (query_rates),
- (query_positions_elems), (query_positions_pads), (do_seek):
- Make correct DV pipeline.
+2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-07-18 Andy Wingo <wingo@pobox.com>
+ * configure.ac:
+ Back to devel -> 0.10.22.1
- * configure.ac (DEFAULT_AUDIOSINK, DEFAULT_AUDIOSRC): Use alsa by
- default. Also because it's the only thing that really works. (This
- is used in the GConf elements).
- Use AS_LIBTOOL_TAGS.
+2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-07-18 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ * common:
+ Install and use pre-commit indentation hook from common
- * gst/playback/gstdecodebin.c: (remove_element_chain):
- * gst/playback/gstplaybin.c: (add_sink):
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
- (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_getcaps), (gst_stream_selector_chain):
- More leak and compile fixes.
+2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-07-18 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * tests/check/libs/rtp.c:
+ Avoid overflows in the padding checks by doing the check slightly differently. Add a unit test to check for correct behaviour.
- * examples/seeking/seek.c: (make_vorbis_theora_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (do_seek), (seek_cb), (stop_seek):
- Updated seek example.
+2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstdecodebin.c: (remove_element_chain), (unlinked):
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
- (queue_out_of_data), (gen_preroll_element), (new_decoded_pad):
- * gst/playback/gstplaybin.c: (add_sink):
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
- (gst_stream_info_set_mute):
- Some refcount leak fixes.
+ * autogen.sh:
+ autogen.sh : Use git submodule
-2005-07-16 Wim Taymans <wim@fluendo.com>
+=== release 0.10.22 ===
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Align samples even if we have roundoff errors in the
- timestamp conversion.
+2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-07-16 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/config.h:
+ Release 0.10.22
+ Original commit message from CVS:
+ Release 0.10.22
- * docs/libs/tmpl/gstringbuffer.sgml:
- * examples/seeking/seek.c: (make_vorbis_theora_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (update_scale), (do_seek):
- Updated seek example.
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event),
- (gst_ogg_demux_loop):
- Push out correct discont values.
-
- * ext/theora/theoradec.c: (theora_dec_src_convert),
- (theora_dec_sink_convert), (theora_dec_src_getcaps),
- (theora_dec_sink_event), (theora_handle_type_packet),
- (theora_handle_header_packet), (theora_dec_push),
- (theora_handle_data_packet), (theora_dec_chain),
- (theora_dec_change_state):
- Better timestamping.
-
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_sink_event), (vorbis_dec_push),
- (vorbis_handle_data_packet), (vorbis_dec_chain):
- * ext/vorbis/vorbisdec.h:
- Better timestamping.
+2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times),
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- Handle syncing on timestamps instead of sample offsets. Make
- use of DISCONT values as described in design docs.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
+ Original commit message from CVS:
+ * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
+ * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
+ * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
+ * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
+ * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
+ Use correct struct alignment everywhere to prevent unaligned
+ memory accesses, resulting in SIGBUS on sparc and probably others.
+ Fixes bug #500833.
+
+2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
+ Forward unknown events upstream to allow latency configuration.
+ Fixes bug #567960.
+
+2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (groups_set_locked_state):
+ Provide the right arguments to a debug line.
+
+2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Don't reset the colorkey when element is reused. Fixes #567511.
+
+2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.21.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.21.3 pre-release
+
+2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c:
+ Store the returned signal id in the right slot when
+ registering the pull-buffer signal.
+ Fixes #567168
+ Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
+
+2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixer.c:
+ Small docs addition to clarify that one really mustn't free
+ the constant GList returned (#566812).
+
+2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add GType for GstRTSPUrl and expose a copy function because we can.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
+ (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ * win32/common/libgstrtsp.def:
+ Add GType for GstRTSPUrl and expose a copy function because we can.
+ API: gst_rtsp_url_copy()
+ Fixes #567027.
+
+2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add plugin dependency for the GIO and GVfs modules.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gio/gstgio.c: (plugin_init):
+ Add plugin dependency for the GIO and GVfs modules.
+ Fixes bug #566876.
+
+2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add plugin dependency for the gnomevfs modules.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ Add plugin dependency for the gnomevfs modules.
+ Fixes bug #566875.
+
+2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstcdda.def:
+ Add new symbol to the list of exported symbols.
+
+2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Fix some comments and docs.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
+ (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
+ (activate_group), (deactivate_group), (groups_set_locked_state),
+ (gst_play_bin_change_state):
+ Fix some comments and docs.
+ Post an error message when we fail to link the selector to the sink.
+ Remove pushing of EOS, this seems unneeded.
+ Lock the state of deactivated groups so that they don't accidentally
+ reactivate when the playbin2 state changes.
+ Reuse uridecodebins.
+ Unlock and relock state of groups when playbin goes to NULL.
+ Fixes #566654.
+ Fixes #566341.
+ * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
+ Only do something in the pad removed callback when we are dealing with
+ our sourcepads because the sinkpads don't have a ghostpad.
+
+2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ Make the GType of GstCDDABaseSrcMode public for bindings.
+ Fixes bug #566837.
+
+2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/libvisual/visual.c: (plugin_init):
+ Use new core API to make registry re-scan the plugin
+ whenever visualisations are added or removed (see #350477).
+
+2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
+
+ gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
+ Original commit message from CVS:
+ Patch by: José Alburquerque <jaalburqu svn gnome org>
+ * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
+ * gst-libs/gst/audio/gstaudioclock.h:
+ Make gst_audio_clock_new use const gchar* to ease the wrapping of
+ C++ bindings. Fixes #566723.
+
+2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add pkg-config files for libgstapp. Fixes bug #566761.
+ Original commit message from CVS:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-app.pc.in:
+ Add pkg-config files for libgstapp. Fixes bug #566761.
+
+2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ Make debug categories static. Use _element_class_set_details_simple().
+
+2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
+ (gst_app_sink_class_init), (gst_app_sink_init),
+ (gst_app_sink_dispose), (gst_app_sink_finalize),
+ (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
+ (gst_app_sink_flush_unlocked), (gst_app_sink_start),
+ (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
+ (gst_app_sink_render), (gst_app_sink_getcaps),
+ (gst_app_sink_set_caps), (gst_app_sink_get_caps),
+ (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
+ (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
+ (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
+ (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
+ (gst_app_sink_pull_buffer)::
+ * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
+ * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
+ (gst_app_src_class_init), (gst_app_src_init),
+ (gst_app_src_flush_queued), (gst_app_src_dispose),
+ (gst_app_src_finalize), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_unlock),
+ (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
+ (gst_app_src_is_seekable), (gst_app_src_check_get_range),
+ (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
+ (gst_app_src_set_caps), (gst_app_src_get_caps),
+ (gst_app_src_set_size), (gst_app_src_get_size),
+ (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
+ (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
+ (gst_app_src_set_latencies), (gst_app_src_set_latency),
+ (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
+ (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
+ * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
+ Move private data into a private instance struct. Add padding to
+ instance and class structures exposed in public headers. Add
+ Since markers to the gtk-doc blurbs (#566750).
+
+2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/app/appsrc_ex.c: Some comments.
+ Original commit message from CVS:
+ * tests/examples/app/appsrc_ex.c: (main):
+ Some comments.
+ When pulling a buffer we can get NULL when the element is EOS, don't try
+ to unref this NULL buffer.
+
+2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
+ Original commit message from CVS:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video.h:
+ Fix up build flags and include statement for the new generated
+ enumtypes files, to fix dist.
+
+2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-app.xml:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/app/Makefile.am:
+ Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
+
+2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_change_state):
+ Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
+ take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
+ this because the async_play method is deprecated and usually not called
+ anymore.
+
+2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
+ Disconnect signal handlers before destroying a previous decodebin so
+ that we don't end up causing deadlocks. Fixes #566586.
+
+2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_init),
+ (gst_audio_test_src_check_get_range),
+ (gst_audio_test_src_set_property),
+ (gst_audio_test_src_get_property):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Add property to control pull/push based scheduling.
+
+2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
+
+ Make the seek and colorkey examples depend on gtk+-x11 as they use
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/examples/seek/Makefile.am:
+ * tests/icles/Makefile.am:
+ Make the seek and colorkey examples depend on gtk+-x11 as they use
+ GDK_WINDOW_XID.
+ Fixes the build with gtk+-quartz.
+
+2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/common/: Add new exports to win32 files.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ * win32/common/libgsttag.def:
+ * win32/common/libgstvideo.def:
+ Add new exports to win32 files.
+
+2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
+ * gst-libs/gst/tag/gsttagdemux.h:
+ Add GType for GstTagDemuxResult enum.
+
+2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
+ Original commit message from CVS:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video.h:
+ Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
+ This will help bindings to use it.
+
+2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/audio/testchannels.c:
+ * win32/MANIFEST:
+ * win32/common/audio-enumtypes.c:
+ (gst_audio_channel_position_get_type),
+ (gst_ring_buffer_state_get_type),
+ (gst_ring_buffer_seg_state_get_type),
+ (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
+ * win32/common/audio-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ * win32/common/multichannel-enumtypes.h:
+ * win32/vs6/grammar.dsp:
+ * win32/vs6/libgstaudio.dsp:
+ * win32/vs7/libgstaudio.vcproj:
+ * win32/vs8/libgstaudio.vcproj:
+ Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
+ audio- in order to wrap all enums declarations of that library.
+ This modification should not matter since that header file is not a
+ public header (it will be included by public headers).
+ Modify win32 crap^Wfiles accordingly.
+
+2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Complete Sebastien's commit from the 13th by exporting the
+ _slave_method_get_type() methods.
+
+2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_init), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_query),
+ (gst_app_src_set_latencies), (gst_app_src_set_latency),
+ (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
+ * gst-libs/gst/app/gstappsrc.h:
+ Add properties and methods to configure and retrieve the min and max
+ latencies.
+
+2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/: Implement URI query. Fixes bug #562949.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
+ (gst_gio_base_src_query):
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_query):
+ Implement URI query. Fixes bug #562949.
+
+2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Add some debug info.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (no_more_pads_cb):
+ Add some debug info.
+ * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
+ (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
+ (gst_play_sink_release_pad):
+ Add some more debug info.
+ Reconfigure the audio chain when we switch between raw and encoded audio
+ in gapless playback.
+
+2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_setcaps):
+ Pause the write thread before deactivating and releasing the ringbuffer
+ to avoid a deadlock when we do gapless playback with different sample
+ rates in playbin2. Fixes #564929.
+
+2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ Make GstAudioSrcSlaveMethod get_type() function non-static
+ as it's public now.
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstnetbuffer.def:
+ Add some missing functions to the list of exported symbols.
+
+2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
+
+ gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
+ Original commit message from CVS:
+ Patch by: Andrew Feren <acferen at yahoo dot com>
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
+ (gst_netaddress_get_address_bytes),
+ (gst_netaddress_set_address_bytes):
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ Make gst_netaddress_get_ip4_address fail for v6 addresses.
+ Make gst_netaddress_get_ip6_address either fail or return the v4
+ address as a transitional v6 address.
+ Add two convenience functions:
+ API: gst_netaddress_get_address_bytes()
+ API: gst_netaddress_set_address_bytes()
+ Fixes #564896.
+
+2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add appsrc and appsink documentation.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
+ Add appsrc and appsink documentation.
+
+2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
+ Original commit message from CVS:
+ * gst/adder/Makefile.am:
+ * gst/adder/gstadder.c:
+ Cleanup variable names to make the adder-loop easier to understand.
+ Also try to use liboil to spee it up, but ifdef it out as it does not
+ make any change for me (Intel pentim M (sse,sse2) please try on other
+ systems).
+
+2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add minimal docs to make the remaining tcp elements show up.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversrc.c:
+ Add minimal docs to make the remaining tcp elements show up.
+ Fixes #564139.
+
+2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Fix example to unref after emiting the push-buffer action.
+ Original commit message from CVS:
+ * examples/app/appsrc-ra.c: (feed_data):
+ * examples/app/appsrc-seekable.c: (feed_data):
+ * examples/app/appsrc-stream.c: (read_data):
+ * examples/app/appsrc-stream2.c: (feed_data):
+ Fix example to unref after emiting the push-buffer action.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
+ (gst_app_src_push_buffer_action):
+ Don't take the ref on the buffer in push-buffer action because it's too
+ awkward for bindings. Fixes #564482.
+
+2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ win32/common/config.h: Update to CVS version.
+ Original commit message from CVS:
+ * win32/common/config.h:
+ Update to CVS version.
+ * win32/common/config.h.in:
+ Hardcode path to plugin install helper exe, just like we hardcode
+ the paths in core. Removes another source of VCS conflicts for
+ people hacking gst-plugins-base on systems with autotools.
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times),
- (gst_ximagesink_show_frame):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
- Correcly convert buffer timestamp to stream time.
+2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-07-16 Wim Taymans <wim@fluendo.com>
+ m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
+ Original commit message from CVS:
+ * m4/Makefile.am:
+ And a couple more .m4 that don't exist anymore with gettext 0.17
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_buffer):
- Timestamp buffers correctly.
+2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstplaybin.c: (gen_video_element):
- Make internal fakesink silent.
+ m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
+ Original commit message from CVS:
+ * m4/Makefile.am:
+ inttypes.m4 hasn't been available since gettext-0.15, and since we now
+ require gettext >= 0.17 ... we can remove it from the list of files to
+ dist.
-2005-07-15 Wim Taymans <wim@fluendo.com>
+2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info),
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
- (gst_ffmpegcsp_get_type), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_get_size),
- (gst_ffmpegcsp_transform_ip), (gst_ffmpegcsp_transform):
- Ported ffmpegcolorspace to basetransform.
+ gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_slave_method_get_type),
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_slave_method_get_type),
+ (gst_base_audio_src_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
+ public API. This is needed for the C++ bindings to be able
+ to use this base classes. Fixes bug #564200, #564206.
+
+2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_handle_event):
+ Remove erroneous gst_buffer_ref().
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Don't forget to unref the buffer once you're done with it.
+
+2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: XRef to GstXOverlay.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ XRef to GstXOverlay.
+
+2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
+ Free the factory array when finalizing.
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
+ Use a GstStaticPadTemplate since the src pad caps are fixed.
+
+2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
+ (gst_vorbis_enc_init):
+ Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
+ pad templates.
+
+2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add mapping for VP6 in avi/riff.
+
+2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
+ Original commit message from CVS:
+ * gst/subparse/samiparse.c: (sami_context_push_state),
+ (sami_context_pop_state), (start_sami_element), (end_sami_element):
+ Some versions of libxml seem to be very picky as to strict formatting
+ of the input and never 'close' the final </body> tag.
+ In order to fix that bad behaviour, we trigger the flushing of
+ remaining data on both </body> and </sami>.
+ Fixes #557365
+
+2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
+ Original commit message from CVS:
+ Patch by: Guillaume Emont <guillaume at fluendo dot com>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinders for MS Word files and OS X .DS_Store files to
+ prevent them to be recognized as MPEG files. Fixes bug #564098.
+
+2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Add some more debug info.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gen_audio_chain),
+ (gst_play_sink_reconfigure):
+ Add some more debug info.
+ Fix linking of just an encoded sink.
+ Handle failure to create a sink chain more gracefully than crashing.
+
+2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ Pushing 10 buffers is enough to run the test.
+
+2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Hook up the SKIP seek flag.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (do_seek), (stop_cb),
+ (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
+ (main):
+ Hook up the SKIP seek flag.
+
+2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (pad_added_cb):
+ Error out with a missing-plugin error when the input-selector was not
+ found.
+ * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
+ Indentation.
+
+2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
+ (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_send_event), (gst_play_sink_change_state):
+ Use G_DEFINE_TYPE.
+ Try to set the selected sink to READY before using it. This will allow
+ for detection of incompatible formats sooner.
+ Don't cause a fatal error when conversion elements are missing but post
+ a missing-element message and a warning instead because things might
+ still link and run fine.
+ Simplyfy the construction of audio and video sink chains.
+
+2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
+ (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
+ Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
+ init from glib.
+
+2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
+
+ gst/: Include glib.h instead of a specific GLib header. Including single
+ Original commit message from CVS:
+ Patch by: Luis Menina <liberforce at freeside dot fr>
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ Include glib.h instead of a specific GLib header. Including single
+ GLib headers is deprecated. Fixes bug #563904.
+
+2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
+
+ gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
+ Original commit message from CVS:
+ 2008-12-09 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
+
+2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c:
+ Fix handling of odd chunks in riff metadata.
+
+2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (gst_volume_class_init),
+ (volume_before_transform), (volume_transform_ip):
+ Use new basetransform vmethod to reconfigure the dynamic properties and
+ any pending volume/mute changes. Fixes #563508.
+
+2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
+ Original commit message from CVS:
+ * configure.ac:
+ First check for "theoraenc theoradec" and if that failed check
+ for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
+ deprecate the latter. Also linking on Windows fails with just "theora"
+ and the version check would fail for the release candidates.
+ Fixes bug #563718.
+
+2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ Add basic docs to decodebin and link to decodebin from decodebin2.
+
+2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester at tester ca>
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Implement gst_rtcp_packet_remove(). Fixes #563174.
+ * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
+ Add unit test for some RTCP functions.
+
+2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
+ Original commit message from CVS:
+ * configure.ac:
+ Apparently AC_CONFIG_MACRO_DIR breaks when using more
+ than one macro directory, reverting last change.
+
+2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
+ Original commit message from CVS:
+ * configure.ac:
+ Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
+ our M4 macros.
+
+2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ sys/: Clear all flags on buffers returned from the image pool.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
+ Clear all flags on buffers returned from the image pool.
+ Fixes #563143
+
+2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
+
+ gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
+ Original commit message from CVS:
+ Patch by: 이문형 <iwings at gmail dot com>
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
+ Don't forget to release the lock again if we bail out because some
+ pad is flushing or we've reached EOS, otherwise things will lock up
+ next time _push_buffer() is called (#562802).
+
+2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
+ Original commit message from CVS:
+ Patch by: Cygwin Ports maintainer
+ <yselkowitz at users dot sourceforge dot net>
+ * autogen.sh:
+ * configure.ac:
+ Require gettext 0.17 because older versions don't mix with libtool
+ 2.2. At build time an older gettext version will still work.
+ Fixes bug #556091.
+
+2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_transform):
- * gst/volume/gstvolume.c: (volume_transform):
- Ported to new API.
+ * ChangeLog:
+ * gst/speexresample/Makefile.am:
+ fix build
+ Original commit message from CVS:
+ fix build
+
+2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Update documentation of speexresample for the new element name.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * gst/speexresample/gstspeexresample.c:
+ Update documentation of speexresample for the new element name.
+
+2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ Update README with the latest diff between the Speex resampler
+ and our copy.
+
+2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c: (plugin_init):
+ Update the debug category from speex_resample to audioresample.
+
+2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Remove audioresample files.
+ Original commit message from CVS:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/buffer.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/debug.c:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/functable.c:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample.h:
+ * gst/audioresample/resample_chunk.c:
+ * gst/audioresample/resample_functable.c:
+ * gst/audioresample/resample_ref.c:
+ * tests/check/elements/audioresample.c:
+ Remove audioresample files.
+
+2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
+ Original commit message from CVS:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ Regenerated for library filename change.
+
+2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/speexresample/gstspeexresample.c: (plugin_init):
+ * gst/speexresample/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/elements/speexresample.c: (setup_speexresample),
+ (GST_START_TEST), (test_pipeline):
+ Rename the moved speexresample to audioresample, integrate into the
+ build system and remove the old audioresample from the build system.
+ Fixes bug #558124, #385061, #346218, #116051.
+
+2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
+ Avoid nasty int overflows after about 12 hours and 25 minutes when these
+ code paths are triggered.
+ A free beer to Håvard Graff for finding this!
+
+2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
+ Original commit message from CVS:
+ Patch by: 이문형 <iwings at gmail dot com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect):
+ A successful gst_poll_wait() doesn't always mean successful connect() on
+ Windows. We should check errors by calling gst_poll_fd_has_error().
+ See #561924.
+
+2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
+ Original commit message from CVS:
+ * tests/check/elements/speexresample.c: (test_pipeline):
+ Make unit test again faster to prevent timeouts with valgrind.
+
+2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ Fix typo in the docs.
+
+2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
+ If no stream was found before receiving EOS, post an error message.
+ Fixes #561924.
+
+2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/: Parse segment events.
+ Original commit message from CVS:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: (gst_theora_enc_init),
+ (theora_buffer_from_packet), (theora_push_packet),
+ (theora_enc_sink_event), (theora_enc_is_discontinuous),
+ (theora_enc_chain):
+ Parse segment events.
+ Pass incomming buffer timestamps to outgoing buffers.
+ Use the running_time to construct the granulepos.
+ Fixes #562163.
+
+2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Fix buffer-duration property.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (activate_group):
+ Fix buffer-duration property.
+
+2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_change_state):
+ Really fix audiosink drain handling by keeping track of the running_time
+ of the last sample.
-2005-07-14 Wim Taymans <wim@fluendo.com>
+2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
- * gst/videotestsrc/Makefile.am:
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get_type),
- (gst_videotestsrc_class_init), (gst_videotestsrc_negotiate),
- (gst_videotestsrc_setcaps), (gst_videotestsrc_getcaps),
- (gst_videotestsrc_init), (gst_videotestsrc_event),
- (gst_videotestsrc_create), (gst_videotestsrc_start),
- (gst_videotestsrc_stop), (gst_videotestsrc_get_times),
- (gst_videotestsrc_set_pattern), (gst_videotestsrc_set_property),
- (gst_videotestsrc_get_property):
- * gst/videotestsrc/gstvideotestsrc.h:
- Make videotestsrc a pushsrc.
-
-2005-07-14 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstfdset.c: (gst_fdset_free):
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_clear), (gst_multifdsink_get_stats),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
- (gst_multifdsink_stop):
- * gst/tcp/gstmultifdsink.h:
- 0.8 backporting.
+ gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c:
+ Add notification of current stream. Add ability to configure buffer
+ sizes.
+ * gst/playback/gsturidecodebin.c:
+ Add ability to configure buffer sizes for streaming mode.
+ Bug #561734.
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
- Also draw image when not from a pool.
+2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-07-14 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Time is already in running_time. Remove base_time handling. Fixes
+ audiosinks not draining and thus chopping some audio in the end.
- * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
- (mute_stream), (silence_stream):
- Small debug additions.
+2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
-2005-07-14 Wim Taymans <wim@fluendo.com>
+ ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ If we're muxing a dirac stream, flush the page after every picture.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose),
- (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_loop):
- Better error recovery, ignore unconnected pads and
- non-fatal errors.
+2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-07-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Add one log message to check for audio_drained. Sync one log message
+ with the condition. Send EOS after draining audio in pull mode.
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstgconf.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * docs/libs/tmpl/gsttuner.sgml:
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get_type),
- (gst_tcpclientsrc_class_init):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get_type),
- (gst_tcpserversrc_class_init):
- * sys/v4l/gstv4lelement.c:
- more autistic cleanliness in functions/names/defines
+2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
+ Use gst_buffer_try_new_and_alloc() and fail properly if the
+ allocation failed. This prevents abort() if downstream elements
+ request an insane amount of memory.
+
+2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
+
+ gst/volume/gstvolume.*: Cleanup volume, define and use default values.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (volume_update_volume), (gst_volume_set_volume),
+ (gst_volume_get_volume), (gst_volume_set_mute),
+ (gst_volume_class_init), (gst_volume_init),
+ (volume_process_double), (volume_process_float),
+ (volume_process_int32), (volume_process_int32_clamp),
+ (volume_process_int24), (volume_process_int24_clamp),
+ (volume_process_int16), (volume_process_int16_clamp),
+ (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
+ (volume_transform_ip), (volume_set_property),
+ (volume_get_property):
+ * gst/volume/gstvolume.h:
+ Cleanup volume, define and use default values.
+ Recalculate new volume and mute setup before processing. Fixes #561789.
+ * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
+ Add controller unit test. Patch by: Jonathan Matthew
+ Fix bogus test that messed with basetransform's internal state.
+
+2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
+ Original commit message from CVS:
+ * tests/check/elements/speexresample.c: (GST_START_TEST):
+ Make the unit test a bit faster to prevent timeouts, especially
+ with valgrind.
+
+2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c:
+ Add jpeg and png image media types to the caps. Fixes #561436.
+
+2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gen_audio_chain):
+ Don't post an error when we can't configure the volume but post a
+ warning instead. Fixes #561780.
+
+2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
+
+ gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
+ Original commit message from CVS:
+ Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ Add a zone plate pattern generator based on BBC R&D Report
+ 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
+ kx2=20 ky2=20 kt=1'.
-2005-07-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * configure.ac:
- make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
- added manually to each Makefile.am so we are sure it goes
- *last* and doesn't add -L flags before linking in libs of our
- own, like, say, internal .la libs, that then accidentally pick
- up the installed copy.
- * docs/libs/Makefile.am:
- * ext/alsa/Makefile.am:
- * ext/cdparanoia/Makefile.am:
- * ext/gnomevfs/Makefile.am:
- * ext/libvisual/Makefile.am:
- * ext/ogg/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- * gst/adder/Makefile.am:
- * gst/audioconvert/Makefile.am:
- * gst/audiorate/Makefile.am:
- * gst/audioscale/Makefile.am:
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/playback/Makefile.am:
- * gst/sine/Makefile.am:
- * gst/subparse/Makefile.am:
- * gst/tags/Makefile.am:
- * gst/tcp/Makefile.am:
- * gst/typefind/Makefile.am:
- * gst/videorate/Makefile.am:
- * gst/videoscale/Makefile.am:
- * gst/videotestsrc/Makefile.am:
- * gst/volume/Makefile.am:
- * sys/v4l/Makefile.am:
- * sys/ximage/Makefile.am:
- * sys/xvimage/Makefile.am:
- adapt properly to this change. This should make sure that
- plugins and libs properly link to the as-yet-uninstalled
- copies of stuff like libgstinterfaces and libgstvideo
+ gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_class_init), (gst_speex_resample_set_property),
+ (gst_speex_resample_get_property):
+ Add a "filter-length" property that maps to the quality values
+ for compatibilty with audioresample.
-2005-07-13 Andy Wingo <wingo@pobox.com>
+2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_stop): Fix a spurious warning.
- (gst_v4lsrc_fixate): Fixate on format as well.
+ gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Fix random fat-fingering making this not compile.
- * sys/xvimage/xvimagesink.c (gst_xvimage_buffer_destroy)
- (gst_xvimagesink_xvimage_new): Ref the xvimagesink while the
- buffer points to it.
- (gst_xvimagesink_check_xshm_calls): Don't use our xvimage buffer,
- rather just doing X calls ourselves. Also fixes a memleak.
+2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
-2005-07-12 Andy Wingo <wingo@pobox.com>
+ gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ If the top-level type of the stream is plain text, don't try to decode
+ it, matching behaviour of decodebin.
+ * gst/playback/gstplaysink.c:
+ If we fail to generate a text chain (e.g. due to missing optional
+ plugins), don't crash.
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_get_property)
- (gst_v4lsrc_set_property, gst_v4lsrc_class_init, gst_v4lsrc_init)
- (gst_v4lsrc_create): Re-add the copy-mode property, default to
- TRUE to avoid deadlocks if an element holds on to our buffers.
+2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
-2005-07-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ Fix win32 build. Oops.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init), (gst_sinesrc_create),
- (gst_sinesrc_set_property), (gst_sinesrc_get_property),
- (gst_sinesrc_start):
- * gst/sine/gstsinesrc.h:
- removing num-buffers property before moving it
+2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
-2005-07-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ Use WSAGetLastError() rather than errno/h_errno on win32.
- * configure.ac:
- use overridable ERROR_CFLAGS
- * docs/libs/gst-plugins-base-libs.types:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
- (gst_alsasink_class_init):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_type),
- (gst_alsasrc_class_init):
- * gst-libs/gst/audio/audio.h:
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
- (gst_audio_filter_base_init), (gst_audio_filter_class_init),
- (gst_audio_filter_link), (gst_audio_filter_init),
- (gst_audio_filter_chain), (gst_audio_filter_set_property),
- (gst_audio_filter_get_property),
- (gst_audio_filter_class_add_pad_templates):
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_get_type),
- (gst_audio_filter_template_base_init),
- (gst_audio_filter_template_class_init),
- (gst_audio_filter_template_init),
- (gst_audio_filter_template_set_property),
- (gst_audio_filter_template_get_property), (plugin_init),
- (gst_audio_filter_template_setup),
- (gst_audio_filter_template_filter),
- (gst_audio_filter_template_filter_inplace):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release), (gst_audioringbuffer_start),
- (gst_audioringbuffer_stop), (gst_audioringbuffer_delay),
- (gst_audio_sink_base_init), (gst_audio_sink_class_init),
- (gst_audio_sink_init), (gst_audio_sink_create_ringbuffer):
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_get_type),
- (gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_start), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audio_src_base_init),
- (gst_audio_src_class_init), (gst_audio_src_init),
- (gst_audio_src_create_ringbuffer):
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_base_init), (gst_base_audio_sink_class_init),
- (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
- (gst_base_audio_sink_get_clock), (gst_base_audio_sink_get_time),
- (gst_base_audio_sink_set_property),
- (gst_base_audio_sink_get_property), (gst_base_audio_sink_setcaps),
- (gst_base_audio_sink_get_times), (gst_base_audio_sink_event),
- (gst_base_audio_sink_preroll), (gst_base_audio_sink_render),
- (gst_base_audio_sink_create_ringbuffer),
- (gst_base_audio_sink_callback), (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_base_init), (gst_base_audio_src_class_init),
- (gst_base_audio_src_init), (gst_base_audio_src_get_clock),
- (gst_base_audio_src_get_time), (gst_base_audio_src_set_property),
- (gst_base_audio_src_get_property), (gst_base_audio_src_fixate),
- (gst_base_audio_src_setcaps), (gst_base_audio_src_get_times),
- (gst_base_audio_src_event), (gst_base_audio_src_create),
- (gst_base_audio_src_create_ringbuffer),
- (gst_base_audio_src_callback), (gst_base_audio_src_change_state):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
- (gst_ring_buffer_class_init), (gst_ring_buffer_init),
- (gst_ring_buffer_dispose), (gst_ring_buffer_finalize),
- (gst_ring_buffer_debug_spec_caps),
- (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
- (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
- (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
- (gst_ring_buffer_start), (gst_ring_buffer_pause),
- (gst_ring_buffer_stop), (gst_ring_buffer_delay),
- (gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
- (wait_segment), (gst_ring_buffer_commit), (gst_ring_buffer_read),
- (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
- (gst_ring_buffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
- (gst_video_sink_class_init), (gst_video_sink_get_type):
- * gst-libs/gst/video/videosink.h:
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_get_type),
- (gst_multifdsink_class_init),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_change_state):
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
- (gst_tcpclientsink_setcaps):
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_getcaps), (gst_ximagesink_setcaps),
- (gst_ximagesink_change_state), (gst_ximagesink_show_frame),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc),
- (gst_ximagesink_send_pending_navigation),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_get_desired_size),
- (gst_ximagesink_class_init), (gst_ximagesink_get_type):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps),
- (gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_send_pending_navigation),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_get_desired_size), (gst_xvimagesink_class_init),
- (gst_xvimagesink_get_type):
- more macro splitting
-
-2005-07-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
- plug a memleak, allows me to import 1479 albums in one go
- in jamboree
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_type_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- fix some format strings
-
-2005-07-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
- (gst_alsasink_set_property), (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
- (gst_alsasrc_set_property), (gst_alsasrc_get_property):
- add device property
-
-2005-07-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (audiocast_register_listener), (audiocast_thread_run),
- (gst_gnomevfssrc_send_additional_headers_callback),
- (gst_gnomevfssrc_received_headers_callback),
- (gst_gnomevfssrc_push_callbacks), (gst_gnomevfssrc_pop_callbacks),
- (gst_gnomevfssrc_get_icy_metadata), (gst_gnomevfssrc_create),
- (gst_gnomevfssrc_get_size):
- add/clean up debugging
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_init):
- cleanups
-
-2005-07-07 Andy Wingo <wingo@pobox.com>
-
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Also fixate the
- framerate. Need to get a handle on when exactly this function is
- called, tho.
+ gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c:
+ Support WMA Lossless properly.
- * sys/v4l/v4lsrc_calls.h:
- * sys/v4l/v4lsrc_calls.c: Remove sync-related stuff.
- (gst_v4lsrc_get_fps_list): Moved here from gstv4lsrc.c.
- (gst_v4lsrc_buffer_new): Totally derive from GstBuffer.
+2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
- * sys/v4l/v4l_calls.h: Cast to V4lElement.
- * sys/v4l/v4l_calls.c: Header loc fixen, don't load mjpeg, all
- v4lelements are sources.
+ gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ Add "colorspec" property, specifying whether to generate BT.601
+ or BT.709 video. This only affects YCbCr values, not RGB, since
+ if you're generating a 709 test pattern, presumably you want
+ 709 RGB primaries, not 601. Also add "smpte75" pattern, which
+ uses 75% colors instead of 100%, since this is often more useful
+ for testing (and also follows the SMPTE EG-1 guideline).
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/gstv4lxoverlay.c:
- * sys/v4l/gstv4ltuner.h:
- * sys/v4l/gstv4ltuner.c: Header loc fixen.
-
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/gstv4lsrc.c: Crucial GPL update. Clean up a bit, port to
- PushSrc/BaseSrc. Removed most sync-related properties, videorate
- or something should handle that. Made a live source.
+2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lelement.c: Derive from GstPushSrc. No more
- signals. Some cleanups.
+ gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c:
+ Add a "sink-caps" property to decodebin like it's done for decodebin2.
+ Fixes #560380.
- * sys/v4l/gstv4lcolorbalance.h: Interface header update.
+2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/gstv4l.c: Don't register v4lelement, or the jpeg/mjpeg
- stuff.
+ gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ Guard against a NULL dereference I somehow encountered -
+ with a FLUSH_STOP arriving either before basetransform _start(),
+ or after _stop().
+ * gst/typefind/gsttypefindfunctions.c:
+ Make sure we never jump backwards when typefinding corrupt mov files.
- * sys/v4l/Makefile.am: Build everything except the jpeg/mjpeg
- stuff.
+2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/Makefile.am (SUBDIRS): Hit the V4L crack pipe.
+ gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ Fix random type causing a docs warning.
-2005-07-07 Wim Taymans <wim@fluendo.com>
+2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/theora/theoradec.c: (theora_get_query_types),
- (theora_dec_src_getcaps), (theora_dec_push):
- * ext/vorbis/vorbisdec.c: (vorbis_get_query_types):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types):
- Remove deprecated/unused query types.
+ sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
+ Original commit message from CVS:
+ * sys/v4l/gstv4l.c:
+ Give it a minimal rank for autovideosrc.
-2005-07-06 Wim Taymans <wim@fluendo.com>
+2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_type),
- (gst_alsasrc_dispose), (gst_alsasrc_base_init),
- (gst_alsasrc_class_init), (gst_alsasrc_init),
- (gst_alsasrc_getcaps), (set_hwparams), (set_swparams),
- (alsasrc_parse_spec), (gst_alsasrc_open), (gst_alsasrc_close),
- (xrun_recovery), (gst_alsasrc_read), (gst_alsasrc_delay),
- (gst_alsasrc_reset):
- * ext/alsa/gstalsasrc.h:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (gst_audioringbuffer_start):
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_get_type),
- (gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
- (gst_audioringbuffer_init), (gst_audioringbuffer_dispose),
- (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release), (gst_audioringbuffer_start),
- (gst_audioringbuffer_stop), (gst_audioringbuffer_delay),
- (gst_audiosrc_base_init), (gst_audiosrc_class_init),
- (gst_audiosrc_init), (gst_audiosrc_create_ringbuffer):
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
- (gst_baseaudiosink_get_time), (gst_baseaudiosink_setcaps),
- (gst_baseaudiosink_preroll), (gst_baseaudiosink_render):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_baseaudiosrc_base_init), (gst_baseaudiosrc_class_init),
- (gst_baseaudiosrc_init), (gst_baseaudiosrc_get_clock),
- (gst_baseaudiosrc_get_time), (gst_baseaudiosrc_set_property),
- (gst_baseaudiosrc_get_property), (gst_baseaudiosrc_fixate),
- (gst_baseaudiosrc_setcaps), (gst_baseaudiosrc_get_times),
- (gst_baseaudiosrc_event), (gst_baseaudiosrc_create),
- (gst_baseaudiosrc_create_ringbuffer), (gst_baseaudiosrc_callback),
- (gst_baseaudiosrc_change_state):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
- (gst_ringbuffer_debug_spec_caps), (gst_ringbuffer_debug_spec_buff),
- (gst_ringbuffer_parse_caps), (gst_ringbuffer_start),
- (gst_ringbuffer_pause), (gst_ringbuffer_stop),
- (gst_ringbuffer_samples_done), (gst_ringbuffer_set_sample),
- (wait_segment), (gst_ringbuffer_commit), (gst_ringbuffer_read),
- (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance):
- * gst-libs/gst/audio/gstringbuffer.h:
- Added audiosource base classes.
- Ported alsasrc, still very basic.
+ gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
+ (plugin_init):
+ Improve typefinding of ISO JPEG2000 mime types.
+
+2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
+ (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.h:
+ Avoid typechecking when we do trivial casts.
+ Move error handling out of the main program flow.
+ Sneak in the display-region caps property, not completely correct yet.
+ Cache the width/height in buffer_alloc instead of parsing it from the
+ caps all the time.
+
+2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (deactivate_group):
+ don't try to unlink the selector sinkpad when we don't have it yet. This
+ can happen if an error occured before the group was complete.
+
+2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
+ (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
+ (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
+ (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
+ (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
+ (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
+ (gst_rtp_buffer_get_extension_data),
+ (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
+ (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
+ (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
+ (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
+ (gst_rtp_buffer_get_payload_type),
+ (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
+ (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
+ (gst_rtp_buffer_set_timestamp),
+ (gst_rtp_buffer_get_payload_subbuffer),
+ (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
+ Avoid expensive type checks we already did as part of the
+ _validate() function that should be called first.
+
+2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
+ (gst_base_rtp_depayload_push_full),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Fix some cases where a newsegment event was not sent.
+
+2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (activate_group):
+ Catch state change errors and stop from the uridecodebin elements
+ instead of trying to continue in vain.
+
+2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/: Wim, you're a bad boy. You don't want people to contact you or what?
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst/h264parse/gsth264parse.c:
+ Wim, you're a bad boy. You don't want people to contact you or what?
+
+2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_callback):
+ Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
+ for the latency to expire, fixes #559567.
+
+2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Change author string after seeing output of gst-inspector.
-2005-07-06 Wim Taymans <wim@fluendo.com>
+2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
+ Don't try to do crazy things when we only have a text pad without a
+ video pad. Fixes #559478.
+
+2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsrc.*: Add is-live property.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_init), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_push_buffer):
+ * gst-libs/gst/app/gstappsrc.h:
+ Add is-live property.
+ Add some more docs.
+
+2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Fix case where we don't have a range for the rates or channels as is the
+ case with truespeech.
+
+2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_update_real_volume),
+ (gst_volume_set_volume), (gst_volume_get_volume),
+ (gst_volume_set_mute), (gst_volume_init), (volume_setup),
+ (volume_transform_ip), (volume_update_mute),
+ (volume_update_volume), (volume_get_property):
+ * gst/volume/gstvolume.h:
+ Keep negotiated state in a separate variable.
+ Protect the volume and mute properties with the object lock.
+ Protect modifying the transform with the transform lock.
+
+2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps):
+ Only convert caps to string when debug is enabled.
+
+2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/: Copy seqnum.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (gst_theora_dec_reset), (theora_dec_src_event),
+ (theora_dec_sink_event), (theora_handle_type_packet):
+ Copy seqnum.
+ Keep events in a pending list, like vorbisdec, instead of trying
+ to construct a segment event ourselves.
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
+ (vorbis_dec_src_event), (vorbis_dec_sink_event):
+ * ext/vorbis/vorbisdec.h:
+ Copy seqnum.
+
+2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
+ (gst_ogg_demux_loop):
+ * ext/ogg/gstoggdemux.h:
+ Copy seqnums around to track playback segments and messages.
+
+2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Don't install static libs for plugins. Fixes #550851 for -bad.
+ Original commit message from CVS:
+ * ext/alsaspdif/Makefile.am:
+ * ext/amrwb/Makefile.am:
+ * ext/apexsink/Makefile.am:
+ * ext/arts/Makefile.am:
+ * ext/artsd/Makefile.am:
+ * ext/audiofile/Makefile.am:
+ * ext/audioresample/Makefile.am:
+ * ext/bz2/Makefile.am:
+ * ext/cdaudio/Makefile.am:
+ * ext/celt/Makefile.am:
+ * ext/dc1394/Makefile.am:
+ * ext/dirac/Makefile.am:
+ * ext/directfb/Makefile.am:
+ * ext/divx/Makefile.am:
+ * ext/dts/Makefile.am:
+ * ext/faac/Makefile.am:
+ * ext/faad/Makefile.am:
+ * ext/gsm/Makefile.am:
+ * ext/hermes/Makefile.am:
+ * ext/ivorbis/Makefile.am:
+ * ext/jack/Makefile.am:
+ * ext/jp2k/Makefile.am:
+ * ext/ladspa/Makefile.am:
+ * ext/lcs/Makefile.am:
+ * ext/libfame/Makefile.am:
+ * ext/libmms/Makefile.am:
+ * ext/metadata/Makefile.am:
+ * ext/mpeg2enc/Makefile.am:
+ * ext/mplex/Makefile.am:
+ * ext/musepack/Makefile.am:
+ * ext/musicbrainz/Makefile.am:
+ * ext/mythtv/Makefile.am:
+ * ext/nas/Makefile.am:
+ * ext/neon/Makefile.am:
+ * ext/ofa/Makefile.am:
+ * ext/polyp/Makefile.am:
+ * ext/resindvd/Makefile.am:
+ * ext/sdl/Makefile.am:
+ * ext/shout/Makefile.am:
+ * ext/snapshot/Makefile.am:
+ * ext/sndfile/Makefile.am:
+ * ext/soundtouch/Makefile.am:
+ * ext/spc/Makefile.am:
+ * ext/swfdec/Makefile.am:
+ * ext/tarkin/Makefile.am:
+ * ext/theora/Makefile.am:
+ * ext/timidity/Makefile.am:
+ * ext/twolame/Makefile.am:
+ * ext/x264/Makefile.am:
+ * ext/xine/Makefile.am:
+ * ext/xvid/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/dshow/Makefile.am:
+ * gst/aiffparse/Makefile.am:
+ * gst/app/Makefile.am:
+ * gst/audiobuffer/Makefile.am:
+ * gst/bayer/Makefile.am:
+ * gst/cdxaparse/Makefile.am:
+ * gst/chart/Makefile.am:
+ * gst/colorspace/Makefile.am:
+ * gst/dccp/Makefile.am:
+ * gst/deinterlace/Makefile.am:
+ * gst/deinterlace2/Makefile.am:
+ * gst/dvdspu/Makefile.am:
+ * gst/festival/Makefile.am:
+ * gst/filter/Makefile.am:
+ * gst/flacparse/Makefile.am:
+ * gst/flv/Makefile.am:
+ * gst/games/Makefile.am:
+ * gst/h264parse/Makefile.am:
+ * gst/librfb/Makefile.am:
+ * gst/mixmatrix/Makefile.am:
+ * gst/modplug/Makefile.am:
+ * gst/mpeg1sys/Makefile.am:
+ * gst/mpeg4videoparse/Makefile.am:
+ * gst/mpegdemux/Makefile.am:
+ * gst/mpegtsmux/Makefile.am:
+ * gst/mpegvideoparse/Makefile.am:
+ * gst/mve/Makefile.am:
+ * gst/nsf/Makefile.am:
+ * gst/nuvdemux/Makefile.am:
+ * gst/overlay/Makefile.am:
+ * gst/passthrough/Makefile.am:
+ * gst/pcapparse/Makefile.am:
+ * gst/playondemand/Makefile.am:
+ * gst/rawparse/Makefile.am:
+ * gst/real/Makefile.am:
+ * gst/rtjpeg/Makefile.am:
+ * gst/rtpmanager/Makefile.am:
+ * gst/scaletempo/Makefile.am:
+ * gst/sdp/Makefile.am:
+ * gst/selector/Makefile.am:
+ * gst/smooth/Makefile.am:
+ * gst/smoothwave/Makefile.am:
+ * gst/speed/Makefile.am:
+ * gst/speexresample/Makefile.am:
+ * gst/stereo/Makefile.am:
+ * gst/subenc/Makefile.am:
+ * gst/tta/Makefile.am:
+ * gst/vbidec/Makefile.am:
+ * gst/videodrop/Makefile.am:
+ * gst/videosignal/Makefile.am:
+ * gst/virtualdub/Makefile.am:
+ * gst/vmnc/Makefile.am:
+ * gst/y4m/Makefile.am:
+ * sys/acmenc/Makefile.am:
+ * sys/cdrom/Makefile.am:
+ * sys/dshowdecwrapper/Makefile.am:
+ * sys/dshowsrcwrapper/Makefile.am:
+ * sys/dvb/Makefile.am:
+ * sys/dxr3/Makefile.am:
+ * sys/fbdev/Makefile.am:
+ * sys/oss4/Makefile.am:
+ * sys/qcam/Makefile.am:
+ * sys/qtwrapper/Makefile.am:
+ * sys/vcd/Makefile.am:
+ * sys/wininet/Makefile.am:
+ * win32/common/config.h:
+ Don't install static libs for plugins. Fixes #550851 for -bad.
+
+2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
+
+ ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
+ Original commit message from CVS:
+ Based on patch by: Matthias Kretz <kretz at kde dot org>
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open),
+ (gst_alsasink_prepare), (gst_alsasink_unprepare),
+ (gst_alsasink_write):
+ Make all access non-blocking so that we can better handle unplugging
+ of usb devices. Fixes #559111
+
+2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien.lespiau gmail com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_write):
+ Make the next call to poll not depend on previous calls to poll with or
+ without reading from the active descriptor. Fixes #544293.
+
+2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_convert_buffer):
+ Add TODO at the top of the file for enabling SSE/ARM specific
+ optimizations and choosing the fastest implementation at runtime.
+ Add g_assert_not_reached() at two places that should really never
+ be reached.
+
+2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Fix format string and arguments.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_check_discont):
+ Fix format string and arguments.
+ * gst/speexresample/resample_sse.h:
+ Add missing file.
+
+2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Add missing headers to Makefile.am.
+ Original commit message from CVS:
+ * gst/speexresample/Makefile.am:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
+ (gst_speex_resample_convert_buffer), (_benchmark_int_float),
+ (_benchmark_int_int), (_benchmark_integer_resampling),
+ (plugin_init):
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/resample.c:
+ * gst/speexresample/speex_resampler_double.c:
+ * gst/speexresample/speex_resampler_float.c:
+ * gst/speexresample/speex_resampler_int.c:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ Add missing headers to Makefile.am.
+ Update copyright, years and my mail address.
+ Benchmark the integer resampling implementation against the
+ float implementation and use the faster one for 8/16 bit integer
+ input. On most recent systems the floating point version is faster.
+
+2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
+
+ gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
+ Original commit message from CVS:
+ Patch by: Nick Haddad <nick at haddads dot net>
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add support for other fourcc codes that are commonly used for
+ 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
+ Fixes #558553.
+
+2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_convert_buffer):
+ The length for the buffer conversion function is the number of
+ audio frames, i.e. we need to multiply it by the number of channels
+ to get the number of values. Also spotted by the unit test after
+ running in valgrind.
+
+2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
+ Original commit message from CVS:
+ * tests/check/elements/speexresample.c: (element_message_cb),
+ (eos_message_cb), (test_pipeline), (GST_START_TEST),
+ (speexresample_suite):
+ Add pipeline unit tests for testing all supported formats with
+ up/downsampling and different in/outrates.
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_push_drain), (gst_speex_resample_process):
+ * gst/speexresample/speex_resampler_wrapper.h:
+ Fix bugs identified by the testsuite.
+
+2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
+ (gst_speex_resample_get_funcs),
+ (gst_speex_resample_transform_size),
+ (gst_speex_resample_convert_buffer),
+ (gst_speex_resample_push_drain), (gst_speex_resample_process):
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ Add support for int8, int24 and int32 input by converting internally
+ to/from int16 or double.
+
+2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
+ Original commit message from CVS:
+ * gst/speexresample/Makefile.am:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
+ (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
+ (gst_speex_resample_init_state), (gst_speex_resample_update_state),
+ (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
+ (_gcd), (gst_speex_resample_transform_size),
+ (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
+ (gst_speex_resample_process), (gst_speex_resample_transform),
+ (gst_speex_resample_query), (gst_speex_resample_set_property):
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/resample.c:
+ * gst/speexresample/speex_resampler.h:
+ * gst/speexresample/speex_resampler_double.c:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ * tests/check/elements/speexresample.c: (setup_speexresample),
+ (test_perfect_stream_instance), (GST_START_TEST),
+ (test_discont_stream_instance):
+ Add support for double samples as input and refactor the usage
+ of the different compilation flavors of the speex resampler.
+
+2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ Return the result of parent_class->event().
- * ext/theora/theoradec.c: (theora_dec_src_getcaps),
- (theora_dec_push), (theora_handle_data_packet):
- Prepare for better timestamp fix later.
+2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.c: Fix the docs.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
+ Fix the docs.
+
+2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
+ (gst_speex_resample_get_unit_size),
+ (gst_speex_resample_push_drain), (gst_speex_resample_event),
+ (gst_speex_resample_check_discont), (gst_speex_resample_process),
+ (gst_speex_resample_transform):
+ * gst/speexresample/gstspeexresample.h:
+ Rewrite timestamp tracking to make it more robust and guarantee
+ a continous stream.
+ * tests/check/Makefile.am:
+ * tests/check/elements/speexresample.c: (setup_speexresample),
+ (cleanup_speexresample), (fail_unless_perfect_stream),
+ (test_perfect_stream_instance), (GST_START_TEST),
+ (test_discont_stream_instance), (live_switch_alloc_only_48000),
+ (live_switch_get_sink_caps), (live_switch_push),
+ (speexresample_suite):
+ Add unit tests for speexresample based on the audioresample unit tests.
+
+2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_get_unit_size),
+ (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
+ (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
+ (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
+ (gst_speex_resample_push_drain), (gst_speex_resample_event),
+ (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
+ (gst_speex_resample_process), (gst_speex_resample_transform),
+ (gst_speex_resample_query), (gst_speex_resample_set_property):
+ * gst/speexresample/gstspeexresample.h:
+ Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
+ instead of GST_DEBUG, ...
+
+2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
+ (gst_speex_resample_process):
+ Fixate to the nearest supported rate instead of the first one.
+
+2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init), (audioresample_fixate_caps):
+ Fixate the rate to the nearest supported rate instead of
+ the first one. Fixes bug #549510.
+
+2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/fixed_arm4.h:
+ * gst/speexresample/fixed_arm5e.h:
+ * gst/speexresample/fixed_bfin.h:
+ * gst/speexresample/fixed_debug.h:
+ * gst/speexresample/fixed_generic.h:
+ * gst/speexresample/resample.c: (compute_func), (main), (sinc),
+ (cubic_coef), (resampler_basic_direct_single),
+ (resampler_basic_direct_double),
+ (resampler_basic_interpolate_single),
+ (resampler_basic_interpolate_double), (update_filter),
+ (speex_resampler_init_frac), (speex_resampler_process_native),
+ (speex_resampler_magic), (speex_resampler_process_float),
+ (speex_resampler_process_int),
+ (speex_resampler_process_interleaved_float),
+ (speex_resampler_process_interleaved_int),
+ (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
+ (speex_resampler_reset_mem):
+ * gst/speexresample/speex_resampler.h:
+ Update Speex resampler with latest version from Speex GIT.
+
+2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ win32/common/libgstaudio.def: Add new symbols.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add new symbols.
+
+2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
+ Attempt to make obfuscated code clearer.
+
+2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Move float endianness conversion macros to core. Second part of bug ##555196.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Move float endianness conversion macros to core. Second part of
+ bug ##555196.
+
+2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/: Don't mark as gtk-doc docs as they aren't public.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.h:
+ Don't mark as gtk-doc docs as they aren't public.
- * gst/audioconvert/gstaudioconvert.c:
- List most accurate caps first
+2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-colorkey.c:
+ Allow setting colorkey if possible. Implement property probe interface
+ for optional X features (autopaint-colorkey, double-buffer and
+ colorkey). Fixes #554533
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
- Use proper pad task function.
+2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
+ Remove useless buffer size assignment. It already has this value.
+
+2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
+ (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
+ (gst_audioringbuffer_stop):
+ Implement a separate activate functions to start monitoring the segments
+ or, in pull mode, pulling in data.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
+ (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
+ (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
+ (gst_base_audio_sink_activate_pull),
+ (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_change_state):
+ Implement pad and element convert query function.
+ Activate the ringbuffer.
+ Use the segment last_stop value as the offset to pull.
+ Use new basesink _do_preroll() method to preroll in the pulling thread.
+ Take appropriate locking in the pulling thread.
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Update some docs.
+
+2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
+ Improve MXF typefinding a bit by searching for a header partition
+ pack instead of just a general partition pack and checking more
+ bytes for valid values.
+
+2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/icles/.cvsignore: update ignore file.
+ Original commit message from CVS:
+ * tests/icles/.cvsignore:
+ update ignore file.
+ * tests/icles/Makefile.am:
+ * tests/icles/test-box.c: (make_pipeline), (main):
+ Add another interactive command line experimentation suite for
+ dynamically boxing/cropping/saling an input video.
+
+2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add methods to more accuratly control the pulling thread of a ringbuffer.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
+ (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Add methods to more accuratly control the pulling thread of a
+ ringbuffer.
+ Add format conversion helper code to the ringbuffer.
+ API: GstRingBuffer:gst_ring_buffer_activate()
+ API: GstRingBuffer:gst_ring_buffer_is_active()
+ API: GstRingBuffer:gst_ring_buffer_convert()
+
+2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
+ (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
+ (gst_audioringbuffer_stop):
+ Signal thread startup earlier so that we can immediatly go into pull
+ mode when we have to and block on preroll.
+
+2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_prepare_read):
+ In pull mode we want the callback to prepull a buffer we can preroll on
+ even when we are not yet playing.
+
+2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_show_frame):
- Fix deadlock when alloc failed.
+ Don't install static libs for plugins. Fixes #550851 for base.
+ Original commit message from CVS:
+ * ext/alsa/Makefile.am:
+ * ext/cdparanoia/Makefile.am:
+ * ext/gio/Makefile.am:
+ * ext/gnomevfs/Makefile.am:
+ * ext/libvisual/Makefile.am:
+ * ext/ogg/Makefile.am:
+ * ext/pango/Makefile.am:
+ * ext/theora/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * gst/adder/Makefile.am:
+ * gst/audioconvert/Makefile.am:
+ * gst/audiorate/Makefile.am:
+ * gst/audioresample/Makefile.am:
+ * gst/audiotestsrc/Makefile.am:
+ * gst/ffmpegcolorspace/Makefile.am:
+ * gst/gdp/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/subparse/Makefile.am:
+ * gst/tcp/Makefile.am:
+ * gst/typefind/Makefile.am:
+ * gst/videorate/Makefile.am:
+ * gst/videoscale/Makefile.am:
+ * gst/videotestsrc/Makefile.am:
+ * gst/volume/Makefile.am:
+ * sys/v4l/Makefile.am:
+ * sys/ximage/Makefile.am:
+ * sys/xvimage/Makefile.am:
+ Don't install static libs for plugins. Fixes #550851 for base.
-2005-07-05 Andy Wingo <wingo@pobox.com>
+2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
+ Set the default blocksize to -1 because we will then use the configured
+ samplesperbuffer to create our output buffer.
+
+2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add mappping for the KMVC (Karl Morton's Video) Codec.
+
+2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
+ Don't forget to advance the offset of what we're matching against, else
+ we end up in a forever loop.
+
+2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
+ Improve typefinding a bit. If we don't have a Unicode charset
+ try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
+
+2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/theora/theoradec.c: Fix build on macosx.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_decode_buffer):
+ Fix build on macosx.
+
+2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
+
+ ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
+ Original commit message from CVS:
+ Based on patch by: Robin Stocker <robin at nibor dot org>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (theora_dec_setcaps), (theora_handle_type_packet),
+ (theora_dec_decode_buffer), (theora_dec_change_state):
+ Parse input caps and make the PAR override the encoded PAR when
+ specified by a container. Fixes #555699.
+
+2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Add some more G_LIKELY
+ Fail when the setcaps function was not called.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_set_outcaps):
+ Propagate return value of setcaps.
+
+2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
+ Original commit message from CVS:
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_class_init), (gst_sub_parse_init),
+ (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
+ (get_next_line), (gst_sub_parse_data_format_autodetect),
+ (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
+ (gst_subparse_type_find):
+ * gst/subparse/gstsubparse.h:
+ Add support for UTF16/UTF32 subtitles as long as the first bytes of
+ the first buffer contain the BOM. This also adds support for other
+ encodings that allow NUL bytes via the encoding property.
+ Fixes bugs #552237 and #456788.
+
+2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
+ Don't drop the last byte of image tags if they're not an URI list.
+ Fixes bug #556066.
+
+2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
+ For looking at the 4th byte we have to get 4 bytes of course
+ and not 3.
+
+2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
+ Improve FLAC-without-headers typefinding by looking at most of the
+ frame header and checking if invalid values are used. Should prevent
+ quite some false positives compared to the old version which only
+ check if the first 14 bits are set.
+
+2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Don't assert on caps==NULL.
- * ext/gnomevfs/gstgnomevfssrc.c:
- * gst/sine/gstsinesrc.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/tcp/gsttcpclientsrc.c: s/BASESRC/BASE_SRC/.
+2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect), (handle_buffer),
+ (gst_sub_parse_change_state):
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c: (GST_START_TEST):
+ Add support for subtitle files with UTF-8 BOM at the beginning
+ by simple stripping it from the first line before passing it
+ to any parsing code. Fixes bug #555257 and playback of files
+ created by Gnome Subtitles.
+
+2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_init),
+ (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
+ (gst_audio_test_src_start), (gst_audio_test_src_stop),
+ (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
+ (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Define the default property values in the usual place.
+ Implement start/stop to reset values correctly.
+ Calculate the sample size only once when we negotiate.
+ Rename some values to make more sense.
+ Keep track of our byte range.
+ Add support for pull based scheduling. Disabled for now until we have
+ the whole stack working.
+ Set the BUFFER_OFFSET correctly.
+
+2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
+ Original commit message from CVS:
+ Based on a patch by: xavierb at gmail dot com
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect):
+ * tests/check/elements/subparse.c: (GST_START_TEST):
+ Make the detection of the used subtitle a bit less strict
+ for srt subtitles. Fixes bug #555607.
+
+2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c:
+ (gst_vorbis_enc_buffer_check_discontinuous):
+ Fix discontinuity detection which was broken by last commit.
+
+2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
+ Original commit message from CVS:
+ * configure.ac::
+ Require core CVS for ghostpad API additions used by decodebin2.
- * sys/v4l/: Port from 0.8.
+2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
- * Many files: Null if we got it....
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Fix debug statements (space between '%' and actual format).
-2005-07-05 Andy Wingo <wingo@pobox.com>
+2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tcp/gsttcpserversink.c (gst_tcpserversink_handle_server_read):
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_client_queue_data):
- Signedness fixes.
+ gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
+ Remove bogus assert, the decodepad could have been created inside an
+ already existing group.
-2005-07-05 Wim Taymans <wim@fluendo.com>
+2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
- * configure.ac:
- * gst/tcp/Makefile.am:
- * gst/tcp/README:
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_get_type),
- (gst_multifdsink_base_init), (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_remove_client_link),
- (is_sync_frame), (gst_multifdsink_handle_client_write),
- (gst_multifdsink_render), (gst_multifdsink_start),
- (gst_multifdsink_stop), (gst_multifdsink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.c: (gst_tcp_host_to_ip),
- (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps),
- (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
- (gst_tcpclientsink_init), (gst_tcpclientsink_setcaps),
- (gst_tcpclientsink_render), (gst_tcpclientsink_start),
- (gst_tcpclientsink_stop), (gst_tcpclientsink_change_state):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get_type),
- (gst_tcpclientsrc_base_init), (gst_tcpclientsrc_class_init),
- (gst_tcpclientsrc_init), (gst_tcpclientsrc_getcaps),
- (gst_tcpclientsrc_create), (gst_tcpclientsrc_start),
- (gst_tcpclientsrc_stop), (gst_tcpclientsrc_unlock):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpplugin.c: (plugin_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get_type),
- (gst_tcpserversrc_base_init), (gst_tcpserversrc_class_init),
- (gst_tcpserversrc_init), (gst_tcpserversrc_finalize),
- (gst_tcpserversrc_create), (gst_tcpserversrc_start),
- (gst_tcpserversrc_stop):
- * gst/tcp/gsttcpserversrc.h:
- * gst/tcp/gsttcpsink.c:
- * gst/tcp/gsttcpsink.h:
- * gst/tcp/gsttcpsrc.c:
- * gst/tcp/gsttcpsrc.h:
- Ported tcp plugins to 0.9.
-
-
-2005-07-05 Andy Wingo <wingo@pobox.com>
-
- * gst/playback/gstplaybasebin.c (fill_buffer):
- message_new_application fixen.
-
- * gst/videoscale/gstvideoscale.c (gst_videoscale_transform_caps):
- Style fix.
-
-2005-07-04 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- Set caps on output buffer.
-
-2005-07-04 Andy Wingo <wingo@pobox.com>
-
- * ext/gnomevfs/gstgnomevfssrc.c
- (gst_gnomevfssrc_received_headers_callback)
- (audiocast_thread_kill, audiocast_thread_run): FORTIFY fixen,
- hopefully.
-
- * gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
- No refcount leakage.
-
- * configure.ac: Enable -Werror.
-
- * ext/theora/theoradec.c (theora_dec_src_getcaps):
- * gst/audioconvert/bufferframesconvert.c
- (buffer_frames_convert_fixate):
- * gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int)
- (gst_audio_convert_fixate):
- * gst/sine/gstsinesrc.c (gst_sinesrc_src_fixate)
- (gst_sinesrc_create): Fixate func changes.
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_buffer_alloc): Unused var.
-
-2005-07-01 Andy Wingo <wingo@pobox.com>
-
- * ext/theora/theoradec.c (theora_dec_src_getcaps): Implement a
- getcaps to do explicit caps. Needs to be done in all decoders,
- possibly via a base class.
-
- * configure.ac (GST_PLUGIN_LDFLAGS): Add videoscale.
-
- * ext/ogg/gstoggdemux.c (gst_ogg_pad_typefind): No need to set
- caps on the sink pad, just rely on the pad template. Also, setting
- ANY caps on a pad is not valid because the caps are not fixed.
-
- * sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc): Set the
- caps on the buffer, and get the width from the desired_caps if
- they're set.
- (gst_ximagesink_renegotiate_size): Implement via setting the
- desired_caps on the ximagesink.
- (gst_ximagesink_setcaps): Only reset the width of the player if it
- wasn't already set. Not sure if this is right.
- (gst_ximagesink_show_frame): Memcpy only for normal buffers.
-
- * sys/ximage/ximagesink.h (desired_caps): New field, is the caps
- that the user wants. NULL unless the window has been resized.
-
- * gst/volume/gstvolume.c (volume_transform): Adapt to
- basetransform refcount changes.
-
-2005-07-01 Andy Wingo <wingo@pobox.com>
+ * ChangeLog:
+ changelog
+ Original commit message from CVS:
+ changelog
+
+2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
+ Original commit message from CVS:
+ 2008-10-08 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
+ target instead of setting it.
+ (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
+ API for a decode pad. The bugfix is that we set the group in
+ activate(), not when the pad was created because it might be NULL
+ then.
+ (gst_decode_group_control_source_pad, gst_decode_group_expose):
+ Update to use the API.
+
+2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
+ Original commit message from CVS:
+ 2008-10-08 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
+ be a subclass of GstGhostPad.
+ (analyze_new_pad): So, when emitting the signals that determine
+ how we do autoplugging, already create the ghost pad and use it as
+ the pad in the signal arguments. This allows applications to make
+ a connection between the pad passed in e.g. autoplug-continue, and
+ the pad passed in new-decoded-pad.
+ (connect_pad, expose_pad): Update to receive the ghosted decode
+ pad in the args, retargetting it as necessary if we have to plug
+ the target pad through a multiqueue.
+ (gst_decode_group_control_source_pad): Adapt to receive an
+ already-ghosted pad that just needs activation, blocking, and
+ drain notification.
+ (sort_end_pads): Adapt for decode pads actually being pads.
+ (gst_decode_group_expose): Adapt for decode pads actually being
+ pads. Rewrite the decode pad names so they appear in order. Adds a
+ new error case if we couldn't set the name.
+ (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
+ logic.
+ (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
+ New API for the decode pad, needed because we shouldn't do these
+ things inside gst_decode_pad_new(), but after.
+ (gst_decode_pad_new): Change to actually make the real pad, and
+ delay the blocking/drainage bits.
+
+2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
+
+ ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
+ Original commit message from CVS:
+ Patch by: Daniel Drake <dsd at laptop dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
+ Unref all buffers when clearing collectpads. Fixes bug #546955.
+
+2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
+
+ ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
+ Original commit message from CVS:
+ Based on a patch by: Klaas <klaas at rivercrew dot net>
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
+ (gst_vorbis_enc_buffer_check_discontinuous),
+ (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
+ * ext/vorbis/vorbisenc.h:
+ Keep track of the upstream segments and use the running time on that
+ segment instead of the buffer timestamp everywhere. Fixes bug #525807.
+
+2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (audio_convert_convert):
+ Prevent overflows with big buffer when calculating the size of
+ the intermediate buffer by using gst_util_uint64_scale() instead of
+ plain arithmetics. Fixes bug #552801.
+
+2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
+
+ ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
+ Original commit message from CVS:
+ Patch by: Pavel Zeldin <pzeldin at gmail dot com>
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
+ (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
+ (gst_clock_overlay_init), (gst_clock_overlay_set_property),
+ (gst_clock_overlay_get_property):
+ * ext/pango/gstclockoverlay.h:
+ API: Add ability to specify format for date/time display by
+ adding a "time-format" property.
+ Fixes bug #554879.
+
+2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
+
+ gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
+ Original commit message from CVS:
+ Patch by: Jan Gerber <j at oil21 dot org>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add FFV1 fourcc to support playback of FFMPEG lossless video
+ in AVI. Fixes bug #555319.
+
+2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
+ Original commit message from CVS:
+ Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Implement skew clock slaving. Fixes #552559.
+
+2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/: Fix include of config.h
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ * gst-libs/gst/audio/testchannels.c:
+ Fix include of config.h
+
+2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
+ Original commit message from CVS:
+ Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
+ (print_media), (gst_sdp_message_dump):
+ Fix parsing of the c= field containing multicast addresses.
+ Fixes #552199.
+ Add the connection info to the session or streams.
+ Fix parsing of the bandwidth.
+ Add debugging for the connections and bandwidths for a media.
+ Add debugging for the bandwidth of the session.
+
+2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_change_state):
+ Configure the next seqnum and timestamp in the state change so that they
+ can be queried soon after.
+
+2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain):
+ Improve debugging of the rtptime.
+
+2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to development -> 0.10.21.1
+ Original commit message from CVS:
+ * configure.ac:
+ Back to development -> 0.10.21.1
+
+2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/videoscale/gstvideoscale.c:
- * gst/videoscale/gstvideoscale.h: Clean up, port to 0.9. Derives
- from BaseTransform, implements a transform_caps. Removed dead code
- including some PAR stuff that was never reached -- should probably
- be added back somehow.
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
-2005-07-01 Andy Wingo <wingo@pobox.com>
+2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/videoscale: Merge David's work from 0.8 branch. Changes to
- come later.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
+ (plugin_init):
+ Add typefinder for MXF.
-2005-06-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * configure.ac:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-libs.types:
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixeroptions.h:
- * ext/alsa/gstalsamixertrack.h:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/colorbalance/.cvsignore:
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/colorbalance/colorbalance-marshal.list:
- * gst-libs/gst/colorbalance/colorbalance.c:
- * gst-libs/gst/colorbalance/colorbalance.h:
- * gst-libs/gst/colorbalance/colorbalance.vcproj:
- * gst-libs/gst/colorbalance/colorbalancechannel.c:
- * gst-libs/gst/colorbalance/colorbalancechannel.h:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/colorbalance.c:
- (gst_color_balance_class_init):
- * gst-libs/gst/interfaces/colorbalance.h:
- * gst-libs/gst/interfaces/interfaces-marshal.list:
- * gst-libs/gst/interfaces/mixer.c: (gst_mixer_class_init):
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/navigation.c:
- * gst-libs/gst/interfaces/tuner.c: (gst_tuner_class_init):
- * gst-libs/gst/interfaces/tuner.h:
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * sys/ximage/Makefile.am:
- * sys/ximage/ximagesink.c:
- * sys/xvimage/Makefile.am:
- * sys/xvimage/xvimagesink.c:
- fold in all interfaces into an interfaces dir, preserving CVS
- history
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
+ (plugin_init):
+ Add typefinder for MXF.
+
+2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ Only build test-colorkey if GTK+ is available.
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Fix build after riff changes.
+=== release 0.10.21 ===
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps),
- (gst_riff_create_iavs_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst-libs/gst/riff/riff-read.h:
- * gst-libs/gst/riff/riff.c: (gst_riff_init):
- Add gst_riff_init() to initialize the debug category, instead
- of plugin_init(). Port riff-media.[ch] from -THREADED to HEAD.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ Release 0.10.21
+ Original commit message from CVS:
+ Release 0.10.21
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init):
- Oops, I shouldn't apply hacks.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init):
- Remove pad_loop function which doesn't work.
+ configure.ac: 0.10.20.4 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.20.4 pre-release
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain):
- Send EOS when deactivating.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (check_queue), (queue_threshold_reached), (queue_out_of_data),
- (gen_preroll_element), (probe_triggered), (mute_stream),
- (silence_stream), (new_decoded_pad), (setup_substreams),
- (set_active_source):
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property),
- (remove_sinks), (add_sink):
- * gst/playback/gststreaminfo.c: (cb_probe), (gst_stream_info_new):
- Change for new probe API.
+ ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
+ Original commit message from CVS:
+ Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
+ * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
+ Set the BOS flag on the BOS packet. Fixes #553244.
-2005-06-29 Wim Taymans <wim@fluendo.com>
+2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
- (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ringbuffer_set_callback):
- Fix compilation error.
- Ringbuffer starts out as not running.
- Free our clock in dispose.
- When releasing the ringbuffer we need to renegotiate so
- clear the pad caps.
+ gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ (gst_rtsp_message_parse_request),
+ (gst_rtsp_message_parse_response):
+ Fix the g_return_val_if_fail() statements.
-2005-06-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
- * autogen.sh:
- * configure.ac:
- * docs/Makefile.am:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-libs-docs.sgml:
- * docs/libs/gst-plugins-libs-sections.txt:
- * docs/libs/gst-plugins-libs.types:
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ringbuffer_set_callback):
- reinstate gtk-doc docs for plugin libs
+ gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Fail to activate if there's insufficient data in the file to be usable,
+ preventing an assertion fail later. Fixes #552960
-2005-06-28 Wim Taymans <wim@fluendo.com>
+2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Commit stuff that should have gone in last week when I made the pre-releases:
+ Original commit message from CVS:
+ Commit stuff that should have gone in last week when I made the pre-releases:
+ 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
+ * configure.ac:
+ 0.10.20.2 pre-release
+ * po/LINGUAS:
+ * po/id.po:
+ * po/pt_BR.po:
+ New translations.
+
+2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/: Recognise Kate subtitle streams (#550582).
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ Recognise Kate subtitle streams (#550582).
+
+2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
+ Remove trailing comma from enum list, which causes problems
+ with -pendantic (#550729).
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_init):
- Removed pad loop function.
+2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-28 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ (gst_property_probe_get_properties),
+ (gst_property_probe_get_property),
+ (gst_property_probe_probe_property),
+ (gst_property_probe_probe_property_name),
+ (gst_property_probe_needs_probe),
+ (gst_property_probe_needs_probe_name),
+ (gst_property_probe_get_values),
+ (gst_property_probe_get_values_name),
+ (gst_property_probe_probe_and_get_values),
+ (gst_property_probe_probe_and_get_values_name):
+ More sanity checks for our second-favourite interface.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
- If we're building a chain we are not in an error case
- when we queue a buffer.
+2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-28 Andy Wingo <wingo@pobox.com>
+ gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ Check for NULL pointer, in the hope that this fixes #532864.
- * *.c: Don't cast to GstObject before reffing/unreffing.
+2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-27 Andy Wingo <wingo@pobox.com>
+ sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
+ No really, the next release is 0.10.21 (fix Since: tags in docs).
- * gst/videotestsrc/gstvideotestsrc.c
- (gst_videotestsrc_activate_push): Activation API changes.
+2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstdecodebin.c (gst_decode_bin_change_state)
- (gst_decode_bin_dispose): Free dynamics in READY->NULL, because
- they have refs on the decodebin.
+ gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
+ Disable a code path that is now called but causes a deadlock for some
+ reason and is unneeded.
- * ext/ogg/gstoggdemux.c (gst_ogg_pad_class_init): Ref the right
- parent class.
- (gst_ogg_pad_typefind): Don't leak a pad ref.
- (gst_ogg_chain_new_stream): gst_object_unref, not g_object_unref.
- (gst_ogg_demux_sink_activate, gst_ogg_demux_sink_activate_push)
- (gst_ogg_demux_sink_activate_pull): Changes for activation API.
+2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-27 Edward Hervey <edward@fluendo.com>
+ sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ Add a "draw-border" property that can be set to false to disable
+ drawing borders.
+ * tests/icles/test-colorkey.c:
+ * tests/icles/Makefile.am:
+ Add new test application for the colorkey handling.
- * ext/theora/theoradec.c: (theora_dec_change_state):
- re-arranged call to parent's state change in order to avoid locks (or
- worse).
+2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
+ This will also be fixed for upcoming gst-ffmpeg release so that once
+ this release of -base is out, it will work with the latest gst-ffmpeg
+ release.
+
+2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Add Truespeech mapping for RIFF formats (AVI/WAV).
+ Fixes #550656
+
+2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
+ Fixes #550638.
+
+2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ * tests/check/elements/subparse.c:
+ Rework last change, so that we build subparse, but just disable the
+ sami parse functionality, if we're configured to not use xml. In the
+ tests only the sami test is disabled now.
-2005-06-26 Edward Hervey <edward@fluendo.com>
+2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- 2nd argument of 'unknow-type' signal is a GstCaps and not a
- GstMiniObject
+ configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
+ Original commit message from CVS:
+ * configure.ac:
+ Disable subparse when xml is disabled. It woundn't work anyway. Fixes
+ test runs.
-2005-06-25 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
- Set the worker thread's running flag to TRUE before starting the
- thread.
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init):
- Catch a failure to add typefind to the bin.
+2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ po/POTFILES.in: Add some more files with strings for translation.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Add some more files with strings for translation.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init), (gst_sinesrc_create),
- (gst_sinesrc_set_property), (gst_sinesrc_get_property),
- (gst_sinesrc_start):
- * gst/sine/gstsinesrc.h:
- add num-buffers and timestamp-offset properties
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_set_property),
- (gst_videotestsrc_get_property):
- add timestamp-offset property
+2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-23 Christian Schaller <uraeus@gnome.org>
+ Use new geo location tags from core. Fixes #481169
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * tests/check/libs/tag.c:
+ Use new geo location tags from core. Fixes #481169
- * configure.ac: add videorate
- * gst-plugins-base.spec.in: add videorate
+2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-06-23 Wim Taymans <wim@fluendo.com>
+ tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c: (setup_audioresample),
+ (fail_unless_perfect_stream), (test_perfect_stream_instance),
+ (test_discont_stream_instance):
+ Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
+ Add debugging for coherence.
- * gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_event), (gst_videorate_chain):
- Fixed videorate, fixating an already fixated caps is not
- an error.
+2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
-2005-06-23 Wim Taymans <wim@fluendo.com>
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
+ Original commit message from CVS:
+ Patch by: Jonathan Matthew <notverysmart gmail com>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinder for PDF documents (which is nice to have, since it's a
+ common format, but also helps prevent false positives). Fixes #549814.
- * ext/ogg/README:
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_set_header_on_caps):
- Buffer on caps is not boxed anymore.
+2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
+ (no_more_pads_cb):
+ Fix nasty race where multiple decodebins could start pushing data before
+ we manage to configure the sinks, resulting in not-linked errors in
+ typical RTSP streaming cases.
+
+2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
+ Since we now call stop, we trigger this code path that causes a deadlock
+ is apparently not needed.
+
+2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
+ (gst_ring_buffer_stop):
+ Also allow the case where the ringbuffer was paused when we try to stop
+ it so that the basesrc stop function is still called.
+
+2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
+
+ sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
+ Original commit message from CVS:
+ Patch by: Mike Ruprecht <cmaiku at gmail dot com>
+ * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
+ Reprobe devices again instead of taking a cached list as new
+ devices could've been plugged in. Fixes bug #549062.
+
+2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
+
+ ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
+ Original commit message from CVS:
+ Patch by: Alessandro Dessina <alessandro nnva org>
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_activate_chain):
+ Don't add pads and activate them for skeleton streams. These are already
+ handled inside oggdemux. Fixes bug #537599.
+
+2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
+ Reset variable so that query and convert fail after going back to
+ READY. Fixes #548898.
+
+2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
+ If a buffer arrives with a timestamp before the timestamp+duration
+ of the previous buffer clip it instead of dropping it completely.
+ Slight improvement for the unfixable bug #548913.
+
+2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ Take the current timestamp instead of timestamp+duration for the offset.
+ This offset will later be used for calculating the timestamp and
+ otherwise vorbisdec will interpolate timestamps wrong if upstream
+ only sends timestamps and no granulepos.
+
+2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/seek.c: Don't crash when having no visualisations.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Don't crash when having no visualisations.
-2005-06-22 Wim Taymans <wim@fluendo.com>
+2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
- * ext/theora/theoraenc.c: (theora_set_header_on_caps):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
- Set buffers on caps as miniobjects and not as boxed.
+ gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
+ check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
+ Fixes #548065.
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * configure.ac:
- back to HEAD
+ gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
+ When cleaning up the caps fields also remove "depth" for the same
+ reason we remove "width".
-=== release 0.9.1 ===
+2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
+ Add Lead H.264 here as well.
- * .cvsignore:
- * NEWS:
- * README:
- * RELEASE:
- * configure.ac:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- updates for release
+2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
-2005-06-09 Andy Wingo <wingo@pobox.com>
+ gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
+ Original commit message from CVS:
+ 2008-08-14 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps): Add Lead H.264 variant.
- * gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
-
-2005-06-09 Andy Wingo <wingo@pobox.com>
+2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/net/Makefile.am:
- Add gstnet to build.
+ gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ When not slaved to another clock also subtract the base_time from our
+ internal clock time to get the running time.
-2005-06-09 Andy Wingo <wingo@pobox.com>
+2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
- * gst-libs/gst/gconf/gconf.c:
- * gst/playback/test.c:
- * gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
- fixes.
+ ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
+ Original commit message from CVS:
+ * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
+ since it has no basis in libtheora.
- * gst/audioconvert/gstaudioconvert.c: RPAD fixes.
+2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/theora/theoraenc.c (theora_enc_chain):
- * ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.
+ gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ Remove double "interface" from doc-string.
+ * gst-libs/gst/interfaces/xoverlay.h:
+ Document interface.
+ * gst-libs/gst/riff/riff.c:
+ Add basic doc blobs.
- * ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
- RealPad.
+2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-02 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ Don't try to build that example anymore.
- * gst-libs/gst/net/Makefile.am:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs.pc.in:
- Added net stuff, version net lib.
+2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-02 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/.cvsignore:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ * gst-libs/gst/audio/make_filter:
+ Move audiofiltertemplate to gst-template.
- * examples/seeking/seek.c: (make_vorbis_theora_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (do_seek):
- Updated seek example.
+2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-02 Andy Wingo <wingo@pobox.com>
+ More docs and shuffling. What can we do with the hundreds of #defines.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ More docs and shuffling. What can we do with the hundreds of #defines.
- * pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
- * pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
- list.
+2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Don't
- remove the typefind, the bin dispose will do it for us. When it's
- removed and unreffed, the signal handler will be disconnected,
- too.
- (unlinked): It's too difficult to disconnect from unlinked
- handlers, as they are on pads not elements. Just punt if the pads
- aren't grandkids of the bin.
+ gst-libs/gst/: Reducing number of dundocumented symbols.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ * gst-libs/gst/tag/gsttagdemux.h:
+ Reducing number of dundocumented symbols.
-2005-06-02 Wim Taymans <wim@fluendo.com>
+2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/README:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_clear_chains):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_handle_data_packet):
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_enc_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_data_packet):
- * gst/audioconvert/bufferframesconvert.c:
- (buffer_frames_convert_chain):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
- (gst_ffmpegcsp_chain):
- * gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_event), (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_activate),
- (gst_videotestsrc_src_query), (gst_videotestsrc_loop):
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
- (gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
- Cleanups and buffer alloc.
-
-2005-05-31 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_delay):
- Don't try to call the delay method when the device is not
- opened.
-
-2005-05-31 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_open):
- Get actual segment size and buffer size after opening
- the device.
-
-2005-05-30 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_clear_chains):
- Also FLUSH upstream, makes the loop function exit faster.
-
- * ext/theora/theoradec.c: (theora_dec_src_query):
- Some more debug info in the query.
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_setcaps):
- Release lock on par error, better error reporting.
-
-2005-05-26 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_chain),
- (gst_ogg_demux_clear_chains), (gst_ogg_demux_change_state):
- Clear chains in READY
- Queue packets until the chain is activated.
-
-2005-05-25 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/audio/audio.c: Fix doc comment syntax.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Fix doc comment syntax.
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ Add more doc-comments and a FIXME: for the signal.
+
+2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
+ (gst_ogg_mux_request_new_pad):
+ * ext/ogg/gstoggmux.h:
+ Don't pretend to support NEWSEGMENT events, instead override the
+ GstCollectPads event function to return FALSE on NEWSEGMENT events
+ and do the normal work for other events.
+ This prevents elements like flacenc to seek to the start and rewrite
+ some data which then results in a broken Ogg packet.
+
+2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
+
+ Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
+ Original commit message from CVS:
+ Patch by: Frederic Crozat <fcrozat@mandriva.org>
+ * ext/alsa/gstalsaplugin.c: (plugin_init):
+ * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
+ * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
+ * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
+ * gst/playback/gstdecodebin.c: (plugin_init):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
+ * gst/playback/gstplayback.c: (plugin_init):
+ * gst/playback/gstqueue2.c: (plugin_init):
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
+ * sys/v4l/gstv4l.c: (plugin_init):
+ Make sure gettext returns translations in UTF-8 encoding rather
+ than in the current locale encoding (#546822).
+
+2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c:
+ Add audio/x-qdm for qtdemux.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
- (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
- (gst_baseaudiosink_set_property), (build_linear_format),
- (debug_spec_caps), (debug_spec_buffer),
- (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
- (gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
- (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
- (gst_ringbuffer_play), (gst_ringbuffer_pause),
- (gst_ringbuffer_stop), (gst_ringbuffer_delay),
- (gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
- (wait_segment), (gst_ringbuffer_commit),
- (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
- (gst_ringbuffer_clear):
- Various small cleanups.
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_change_state):
- * gst/subparse/gstsubparse.c: (gst_subparse_chain):
- No need to take the locks anymore.
-
-2005-05-25 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (gst_decode_bin_dispose), (try_to_link_1), (get_our_ghost_pad),
- (remove_element_chain), (no_more_pads), (unlinked), (close_link),
- (type_found):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_dispose),
- (group_destroy), (group_commit), (queue_overrun),
- (gen_preroll_element), (no_more_pads), (preroll_unlinked),
- (mute_stream), (new_decoded_pad), (setup_substreams),
- (setup_source), (mute_group_type), (set_active_source),
- (gst_play_base_bin_change_state):
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
- (gen_video_element), (gen_text_element), (gen_audio_element),
- (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_info_new),
- (gst_stream_info_dispose), (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c: (gst_stream_selector_chain):
- Some playbin cleanups mostly refcounting sloppyness.
-
-2005-05-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
- Work with streaming input.
-
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
- (gst_ffmpegcsp_chain), (gst_ffmpegcsp_change_state):
- No need to take the STREAM lock anymore.
+ ext/vorbis/vorbisdec.c: Do not leak old taglist.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c:
+ Do not leak old taglist.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose),
- (gst_ogg_pad_typefind), (gst_ogg_pad_submit_packet),
- (gst_ogg_chain_new_stream), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_chain), (gst_ogg_demux_loop),
- (gst_ogg_demux_sink_activate):
- * ext/theora/theoradec.c: (theora_dec_src_event),
- (theora_handle_comment_packet), (theora_dec_chain),
- (theora_dec_change_state):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_data_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- Remove STREAM locks as they are taken in core now.
- Never set bogus granulepos on vorbis/theora.
- Fix leaks in theoradec tag parsing.
+ tests/icles/test-scale.c: Include <stdlib.h> for atoi().
+ Original commit message from CVS:
+ * tests/icles/test-scale.c:
+ Include <stdlib.h> for atoi().
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Fix memleaks, GST_BUFFER_DATA() is not freed.
+ gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
+ Original commit message from CVS:
+ 2008-08-04 Andy Wingo <wingo@pobox.com>
+ * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
+ documentation fix.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Open non-blocking, set to blocking mode afterwards to avoid
- lockups when audio device is busy.
+ gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Cleanup lots of empty lines that came from gst-indent going havoc
+ before I added the INDENT_ON/OFF marker some time agao.
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
- This can't be good.
+ Bump requirement to latest core and use new tag for riff formats.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/riff/riff-read.c:
+ Bump requirement to latest core and use new tag for riff formats.
+ Needed for #520694.
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_chain), (gst_audio_convert_link_src),
- (gst_audio_convert_setcaps):
- Implement instant setup switching.
+ tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
+ Original commit message from CVS:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/dynamic/codec-select.c: (make_encoder),
+ (make_pipeline), (do_switch), (my_bus_callback), (main):
+ Add example app that dynamically switches between 3 'encoders'.
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Fix missing unlock.
- * gst/playback/gstplaybin.c: (add_sink):
- First add, then link (otherwise pad link fails).
+ gst/playback/gstplaysink.c: Add some more comments.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
+ Add some more comments.
-2005-05-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
- * examples/Makefile.am:
- fix buildbot (make distcheck)
+ gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
+ (gst_video_test_src_create):
+ Discard buffers of the wrong size after renegotiation, this is perfectly
+ possible with things like capsfilter that could suggest caps changes
+ upstream without knowing the size of the buffer.
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybin.c: (gen_vis_element):
- Remove some wrong code. Doesn't work yet.
+ tests/icles/: Add dynamic rescaling tests for the new basetransform.
+ Original commit message from CVS:
+ * tests/icles/.cvsignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-scale.c: (make_pipeline), (main):
+ Add dynamic rescaling tests for the new basetransform.
-2005-05-19 Wim Taymans <wim@fluendo.com>
+2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/net/README:
- * gst-libs/gst/net/gstnetbuffer.c: (gst_netbuffer_get_type),
- (gst_netbuffer_class_init), (gst_netbuffer_init),
- (gst_netbuffer_finalize), (gst_netbuffer_copy),
- (gst_netbuffer_new), (gst_netaddress_set_ip4_address),
- (gst_netaddress_set_ip6_address), (gst_netaddress_get_net_type),
- (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address):
- * gst-libs/gst/net/gstnetbuffer.h:
- Added buffer subclass to store extra to/from addresses for
- network sources/sinks.
+ gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Dist recently-added gstfastrandom.h.
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
- Don't lock an unassigned variable.
+ sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ Fix a "may be used uninitialized in this function" which weirdly only
+ appears on macosx (?).
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstplaybasebin.c: (gen_preroll_element):
- Increase buffer for video, decrease buffer for other media types.
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- Change names for debugging purposes.
+ gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Adding acid chunk for tempo and loop information.
-2005-05-18 Wim Taymans <wim@fluendo.com>
+2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
- (gst_ffmpegcsp_chain):
- Enable buffer alloc passthrough if the source and dest
- formats are the same.
-
-2005-05-17 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
- (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
- (gst_ogg_demux_chain_unlocked):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
- (gst_audio_convert_fixate), (gst_audio_convert_change_state):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_init),
- (gst_ffmpegcsp_bufferalloc), (gst_ffmpegcsp_chain),
- (gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
- (gst_ffmpegcsp_get_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
- (gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_imagepool_clear),
- (gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id):
- Leak fixes in oggdemux.
- Some cleanups in audioconvert.
- Make passthrough work along with buffer_alloc etc.
- Make buffer_alloc and buffer recycling actually work in
- xvimagesink.
-
-2005-05-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/subparse/gstsubparse.c: (parse_subrip), (parse_mpsub):
- make the compiler happy
-
-2005-05-17 Wim Taymans <wim@fluendo.com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
- (gst_xvimage_buffer_init), (gst_xvimage_buffer_class_init),
- (gst_xvimage_buffer_get_type), (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_imagepool_clear), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_change_state), (gst_xvimagesink_show_frame),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_xwindow_id):
- * sys/xvimage/xvimagesink.h:
- Port xvimagesink to new MiniObject.
+ sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
+ Original commit message from CVS:
+ * sys/xvimage/Makefile.am:
+ floor() needs linking to $(LIBM).
-2005-05-17 Wim Taymans <wim@fluendo.com>
+2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
- (gst_audiofilter_chain):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
- Fix passthrough in ffmpegcolorspace.
- Fix memset in audiosink on wrong memory.
-
-2005-05-16 David Schleef <ds@schleef.org>
-
- * gst/playback/gststreaminfo.c: (cb_probe): Port from GstData
- to GstMiniObject.
-
-2005-05-16 David Schleef <ds@schleef.org>
-
- Port from GstData to GstMiniObject.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_collected):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- * ext/theora/theoradec.c: (theora_handle_comment_packet),
- (theora_handle_data_packet):
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_set_header_on_caps), (theora_enc_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_handle_comment_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
- * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_buffer):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
- (mute_stream), (silence_stream):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
- * gst/volume/gstvolume.c: (volume_transform):
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
- (gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
- (gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
- (gst_ximagesink_buffer_alloc):
- * sys/ximage/ximagesink.h:
+ ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Aggregate short reads and add some comments and debug logging.
+ Fixes #537380
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (fill_buffer), (check_queue), (queue_threshold_reached),
- (queue_out_of_data):
- * gst/playback/gstplaybasebin.h:
- Post buffer-fullness on the bus.
+ gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ Fix property doc markup (its not a signal).
+ * sys/xvimage/xvimagesink.c:
+ Add since tag for new proeprties (also add sice tags fro the last two
+ other additions).
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (try_to_link_1):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (group_commit), (probe_triggered), (setup_source),
- (gst_play_base_bin_change_state):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (remove_sinks), (setup_sinks),
- (gst_play_bin_change_state):
- Move setup_output_pads into a virtual function, remove
- group-switch (no longer needed) and redirect (handled by bus
- now) signals.
-
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_finalize),
- (get_active_group), (get_building_group), (group_destroy),
- (group_commit), (check_queue), (queue_overrun),
- (queue_threshold_reached), (queue_out_of_data),
- (gen_preroll_element), (remove_groups), (unknown_type),
- (add_element_stream), (no_more_pads), (probe_triggered),
- (preroll_unlinked), (new_decoded_pad), (setup_subtitle),
- (setup_substreams), (setup_source), (finish_source),
- (prepare_output), (muted_group_change_state),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (gst_play_base_bin_change_state):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (gst_play_bin_set_property),
- (gen_video_element), (gen_text_element), (gen_audio_element),
- (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
- (gst_play_bin_change_state):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
- (cb_probe), (gst_stream_info_new), (gst_stream_info_dispose),
- (stream_info_change_state), (gst_stream_info_set_mute),
- (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- * gst/playback/gststreamselector.c: (gst_stream_selector_init),
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_getcaps),
- (gst_stream_selector_get_linked_pads),
- (gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
- * gst/playback/gststreamselector.h:
- Rough port of playbin. Needs some more work, but is mostly done,
- and uses a few locks in important places, which should make stuff
- like chain-switches clean. Still uses GST_STATE() in a few places,
- which isn't all that good an idea, subtitles/elements disabled
- because no elements to test with and thus probably broken, query
- and event handling moved to GstBin, internal thread removed
- alltogether because the pipeline does that for us now. Can play
- Ogg/Vorbis files. Haven't tested anything else yet.
-
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
- Do no-more-pads (needed for autoplugging).
-
-2005-05-10 Andy Wingo <wingo@pobox.com>
-
- * ext/vorbis/vorbisdec.c (vorbis_handle_comment_packet): Post a
- message to the bus with the tags. Still not sent downstream tho.
-
- * gst/playback/gstdecodebin.c (remove_element_chain): Unref after
- get_parent.
- (remove_element_chain): Use OBJECT_PARENT instead of get_parent to
- avoid refcounting hassles.
-
-2005-05-09 Andy Wingo <wingo@pobox.com>
+ sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ Add autofill/colorkey properties. Fixes #538656.
- * gst/volume/Makefile.am:
- * gst/volume/demo.c
- * gst/volume/gstvolume.h
- * gst/volume/gstvolume.c: Port to 0.9 API, derive from
- basetransform. Probably need an audio filter base class.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
- (gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
- (gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
- (gst_vorbisenc_chain):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- Make caps writable before writing to it.
- Fix negotiation in audioconvert some more.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_event), (gst_videorate_chain):
- Better negotiation.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_blank_data), (gst_videorate_init),
- (gst_videorate_event), (gst_videorate_chain),
- (gst_videorate_change_state):
- Port videorate, do a better job at negotiation while we're at
- it.
-
-2005-05-09 Jan Schmidt <thaytan@mad.scientist.com>
+2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
- * configure.ac:
- Disable libvisual
+ sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Fix rounding errors when converting colorbalance values
+ between hardware and object property ranges. Partial
+ fix for #537889, however, there still seems to be a small
+ drift problem that could be totem's fault.
- * examples/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- Fixups for missing variables.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_theora_pipeline),
- (make_vorbis_theora_pipeline), (make_avi_msmpeg4v3_mp3_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (update_scale), (play_cb), (pause_cb), (stop_cb), (main):
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
- (gst_ogg_pad_query_types), (gst_ogg_pad_src_query),
- (gst_ogg_pad_typefind), (gst_ogg_demux_init),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
- (gst_ogg_demux_read_end_chain), (gst_ogg_demux_sink_activate):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_push_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_collected), (gst_ogg_mux_change_state):
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_sink_query), (gst_ogm_parse_chain):
- * ext/theora/theoradec.c: (gst_theora_dec_init), (_inc_granulepos),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_sink_query),
- (theora_dec_src_event), (theora_dec_sink_event),
- (theora_handle_comment_packet), (theora_handle_type_packet),
- (theora_handle_header_packet), (theora_handle_data_packet),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_sink_query), (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
- (vorbis_handle_type_packet), (vorbis_handle_header_packet),
- (copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
- (gst_vorbisenc_sink_query), (gst_vorbisenc_init),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_chain):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_query):
- * gst/playback/test3.c: (update_scale):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_setcaps),
- (gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start):
- * gst/subparse/gstsubparse.c: (gst_subparse_init):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
- (gst_videotestsrc_src_query):
- * gst/videotestsrc/videotestsrc.c: (paint_hline_I420),
- (paint_hline_Y41B), (paint_hline_Y42B), (paint_hline_Y800),
- (paint_hline_YUV9):
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
- Port to new query API.
- Updated seek.
- Cleanups in x[v]imagesink
-
-2005-05-09 Andy Wingo <wingo@pobox.com>
+2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/alsa/gstalsasink.h:
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnomevfssrc_get_icy_metadata):
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek)
- (gst_ogg_demux_read_chain, gst_ogg_demux_read_end_chain)
- * ext/theora/theoradec.c (theora_dec_src_query)
- (theora_dec_src_event, theora_dec_sink_event)
- (theora_handle_comment_packet, theora_handle_data_packet):
- * ext/theora/theoraenc.c (theora_enc_chain):
- * ext/vorbis/vorbisdec.c (vorbis_dec_src_event)
- (vorbis_dec_sink_event, vorbis_handle_comment_packet):
- * gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
- * gst/typefind/gsttypefindfunctions.c (mp3_type_find)
- (qt_type_find):
- * gst/videotestsrc/videotestsrc.c (paint_setup_I420)
- (paint_setup_YV12, paint_setup_YUY2, paint_setup_UYVY)
- (paint_setup_YVYU, paint_setup_IYU2, paint_setup_Y41B)
- (paint_setup_Y42B, paint_setup_Y800, paint_setup_IMC1)
- (paint_setup_IMC2, paint_setup_IMC3, paint_setup_IMC4)
- (paint_setup_YVU9, paint_setup_YUV9, paint_setup_xRGB8888)
- (paint_setup_xBGR8888, paint_setup_RGBx8888)
- (paint_setup_BGRx8888, paint_setup_RGB888, paint_setup_BGR888)
- (paint_setup_RGB565, paint_setup_xRGB1555):
- * gst/videotestsrc/videotestsrc.h:
- * sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c (gst_xvimagesink_get_xv_support)
- (gst_xvimagesink_setcaps, gst_xvimagesink_buffer_alloc):
- GCC4 fixes.
-
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_find_chains): Use the new
- gst_pad_query_position. Fixes oggdemux.
-
-2005-05-08 David Schleef <ds@schleef.org>
-
- * configure.ac: Require liboil.
- * gst/videotestsrc/gstvideotestsrc.c: Fix up liboil calls, add
- a few more.
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
+ ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
+ Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
+ This fixes a critical warning.
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
- Well, unreffing a buffer right before pushing it is asking
- for trouble..
+ ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Allow muxing of CELT into Ogg streams.
-2005-05-06 Christian Schaller <uraeus@gnome.org>
+2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * pkgconfig/gstreamer-libs.pc.in: add missing library calls
+ gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
+ (plugin_init):
+ Add simple typefinder for the CELT codec (www.celt-codec.org).
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
- * gst/sine/Makefile.am:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get_type),
- (gst_sinesrc_class_init), (gst_sinesrc_init),
- (gst_sinesrc_src_fixate), (gst_sinesrc_setcaps),
- (gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start),
- (gst_sinesrc_update_freq):
- * gst/sine/gstsinesrc.h:
- * gst/tcp/gstmultifdsink.c:
- * sys/xvimage/xvimagesink.c:
- Fixed negotiation wrt _peer_get_caps()
- Some cleanups.
+ ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
+ Original commit message from CVS:
+ Patch by: Jan Gerber <j at oil21 dot org>
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
+ Fix calculation of the start time from skeleton streams.
+ Fixes bug #530068.
+2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-06 Wim Taymans <wim@fluendo.com>
+ tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
- (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
- (gst_baseaudiosink_set_property), (build_linear_format),
- (debug_spec_caps), (debug_spec_buffer),
- (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
- (gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
- (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
- (gst_ringbuffer_play), (gst_ringbuffer_pause),
- (gst_ringbuffer_stop), (gst_ringbuffer_delay),
- (gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
- (wait_segment), (gst_ringbuffer_commit),
- (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
- (gst_ringbuffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- Make the base audiosink return an error when there is no
- audiobuffer negotiated.
+2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioquantize.c:
+ (gst_audio_quantize_setup_dither),
+ (gst_audio_quantize_free_dither):
+ * gst/audioconvert/gstfastrandom.h:
+ Implement a linear congruential generator as pseudo random number
+ generator for the dither noise. This is about 2 times faster than
+ using GLib's mersenne twister. Also this uses only integer math for
+ generating integers while GLib internally uses floating point math.
+
+2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
+
+ configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
+ Original commit message from CVS:
+ * configure.ac:
+ Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
-2005-05-06 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * ext/Makefile.am:
- Disable cdparanoia until someone ports it!
+ gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien.lespiau gmail com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
+ Use GST_STR_NULL to avoid crashes with libcs that don't
+ like NULL strings in printf args (such as the win32 one).
+ Fixes #544306.
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
+ Oops - set the size of the image used for probing back to 1x1, for
+ consistency with ximagesink
+
+2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: it's not legal to ask the
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
+ (gst_ximagesink_ximage_new):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
+ (gst_xvimagesink_xvimage_new):
+ Apparently on Solaris and OS/X (at least), it's not legal to ask the
+ X server to attach to a shared memory segment after we've deleted it,
+ with the result that MIT-SHM is disabled. Instead, remove it only after
+ X succeeds in attaching too.
+
+2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Add 'ticks', a 1/30 second sine wave pulse every second.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_sink_activate):
- And revert after wingo's revert.. sigh..
+2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
-2005-05-05 Andy Wingo <wingo@pobox.com>
+ gst-libs/gst/video/video.c: Revert ABI change.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: Revert ABI change.
- * gst/audiorate/gstaudiorate.c (gst_audiorate_class_init): Pacify
- GObject.
- * configure.ac: Return audiorate and subparse from the ghetto.
- Re-enable -Wall -Werror.
- * gst/subparse/gstsubparse.c:
- * gst/subparse/gstsubparse.h: Port to 0.9. Can operate loop-based
- or chain-based. Cleaned up a bit. Not tested.
-
-2005-05-05 Christian Schaller <christian@fluendo.com>
-
- * Makefile.am: remove stuff that is not building
- * configure.ac: remove stuff that is not building
- * examples/Makefile.am: remove stuff that is not building
- * ext/alsa/gstalsasink.c: add alsa/ before the alsalib.h file
- * ext/alsa/gstalsasink.h: add alsa/ before the alsalib.h file
- * sys/Makefile.am: remove stuff that is not building
- * testsuite/Makefile.am: remove stuff that is not building
-
-2005-05-05 Andy Wingo <wingo@pobox.com>
-
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
- * gst-libs/gst/tag/gstvorbistag.c:
- (gst_tag_list_from_vorbiscomment_buffer), (gst_vorbis_tag_chain):
- * gst/adder/gstadder.h:
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_one_other):
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_setcaps),
- (gst_audiorate_init), (gst_audiorate_chain):
- * gst/playback/gstplaybasebin.c: (setup_source):
- * gst/playback/test3.c: (update_scale):
- Some GCC4 fixes
-
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po: Foo
+2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-05-05 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make it impossible to have NULL caps at the point where we set
+ framerate and other things. Also don't return immediately for "3ivd"
+ video and let framerate, etc be set. Might fix bug #542508.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_setcaps), (gst_audio_convert_fixate),
- (gst_audio_convert_change_state), (gst_audio_convert_channels):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
- (gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
- (gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
- (gst_videotestsrc_init), (gst_videotestsrc_loop):
- Don't ignore _push() return values.
- Make sure no processing is done when shutting down.
- Videotestsrc pad activation fix.
+2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-05-05 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
+ Video format can also be conveniently determined from (many)
+ non-fixed caps.
- * gst/adder/Makefile.am:
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_class_init), (gst_adder_init),
- (gst_adder_request_new_pad), (gst_adder_collected),
- (gst_adder_change_state):
- * gst/adder/gstadder.h:
- Ported adder as an example of a mixer element using
- collect pads. Needs more negotiation work.
-
-2005-05-05 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (_inc_granulepos),
- (theora_dec_src_event), (theora_dec_sink_event),
- (theora_handle_comment_packet), (theora_handle_type_packet),
- (theora_handle_header_packet), (theora_handle_data_packet),
- (theora_dec_chain):
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (gst_theora_enc_init), (theora_enc_sink_setcaps),
- (theora_push_buffer), (theora_push_packet),
- (theora_enc_sink_event), (theora_enc_chain),
- (theora_enc_change_state), (theora_enc_set_property),
- (theora_enc_get_property):
- Added stream lock to decoder so that we can serialize
- the discont event.
- More theoraenc porting, recover from errors, do clean
- shutdown.
-
-2005-05-05 Wim Taymans <wim@fluendo.com>
+2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/Makefile.am:
- * ext/ogg/README:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_pad_submit_packet), (gst_ogg_demux_sink_activate),
- (gst_ogg_print):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_push_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_collected),
- (gst_ogg_mux_change_state):
- Ported ogg muxer.
-
-2005-05-05 Wim Taymans <wim@fluendo.com>
+ gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreamselector.c:
+ First stab at integrating DVD subpicture overlay into
+ playbin. Successfully plugs and plays, but the queues need
+ shrinking - 3 seconds of video is too much buffering.
- * docs/design-audiosinks.txt:
- * gst-libs/gst/audio/TODO:
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
- (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
- (gst_baseaudiosink_set_property), (build_linear_format),
- (debug_spec_caps), (debug_spec_buffer),
- (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
- (gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
- (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_play),
- (gst_ringbuffer_pause), (gst_ringbuffer_stop),
- (gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
- (gst_ringbuffer_set_sample), (wait_segment),
- (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
- (gst_ringbuffer_advance), (gst_ringbuffer_clear):
- More work on the audiosink, mostly debugging and a race in
- shutdown.
-
-2005-04-28 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_sink_activate):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
- (vorbis_dec_src_query), (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
- (vorbis_handle_type_packet), (vorbis_handle_header_packet),
- (copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain):
- Don't crap out when seeking back to position 0.
-
-2005-04-28 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
- (make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
- (make_vorbis_pipeline), (make_vorbis_theora_pipeline),
- (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
- (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline):
- Make audio sink configurable, use alsasink as default.
-
-2005-04-28 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
- (vorbis_handle_type_packet), (vorbis_handle_header_packet),
- (copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- * ext/vorbis/vorbisdec.h:
- Refactor, use STREAM_LOCK.
+2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-04-28 Wim Taymans <wim@fluendo.com>
+ gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ Remove now obsolete note in the docs.
- * ext/theora/theoradec.c: (_inc_granulepos),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_type_packet), (theora_handle_header_packet),
- (theora_handle_data_packet), (theora_dec_chain),
- (theora_dec_change_state):
- Refactor a bit, use STREAM_LOCK.
+2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbistag.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstqueue2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section
+ titles. Drop mentining that all our example pipelines are "simple"
+ pipelines.
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbistag.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstqueue2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section
+ titles. Drop mentining that all our example pipelines are "simple"
+ pipelines.
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_get_caps),
- (gst_alsa_fixate_to_mimetype), (gst_alsa_fixate_field_nearest_int),
- (gst_alsa_link), (gst_alsa_close_audio):
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
- (gst_alsasink_dispose), (gst_alsasink_base_init),
- (gst_alsasink_class_init), (gst_alsasink_init),
- (gst_alsasink_getcaps), (set_hwparams), (set_swparams),
- (alsasink_parse_spec), (gst_alsasink_open), (gst_alsasink_close),
- (xrun_recovery), (gst_alsasink_write), (gst_alsasink_delay),
- (gst_alsasink_reset):
- * ext/alsa/gstalsasink.h:
- Implement alsasink with simple open/write/close API.
- Make alsa dir build by disabling compilation of code.
+2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
+ Original commit message from CVS:
+ * tests/examples/seek/Makefile.am:
+ Fix out of tree build by adding all required CFLAGS.
+
+2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_raw_queue):
+ And ref the pad before returning it again when linking to the queue
+ failed. Otherwise we will unref the pad twice later and things break.
+
+2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_raw_queue):
+ If linking the raw pad with a queue fails, try it without a queue
+ instead of failing completely. This should never happen.
+
+2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
+
+ gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
+ Original commit message from CVS:
+ Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
+ * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
+ Add a queue after a demuxer if the demuxer outputs raw data. This was
+ done before only for non-raw data but is required in this case too.
+ Fixes bug #540215.
+ decodebin2 doesn't have this issue because all streams of a group
+ go through multiqueue.
+
+2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Makes libgstsdp compile with mingw32 by defining the right WINVER so
+ that getaddrinfo() can be used. Fixes #541358.
+
+2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_init),
+ (gst_video_test_src_set_property),
+ (gst_video_test_src_get_property), (gst_video_test_src_create):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Cleanups, use default property values as defines.
+ Add property to enable/disable peer buffer allocation.
+
+2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
+ * tests/check/pipelines/streamheader.c: (streamheader_suite):
+ Enable unit tests on PPC again as the bugs are now fixed.
+
+2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
+ Fixes bug #540351.
+
+2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_get_unit_size):
+ Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
+ it on other formats. Also adjust the unit size only for that format
+ to not include the palette. Fixes bug #540497.
+
+2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/audio.h:
- * gst-libs/gst/audio/audioclock.c:
- * gst-libs/gst/audio/audioclock.h:
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_get_type),
- (gst_audio_clock_class_init), (gst_audio_clock_init),
- (gst_audio_clock_new), (gst_audio_clock_get_internal_time):
- * gst-libs/gst/audio/gstaudioclock.h:
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
- (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
- (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property),
- (build_linear_format), (debug_spec_caps), (debug_spec_buffer),
- (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
- (gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
- (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_init), (gst_ringbuffer_finalize),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_play),
- (gst_ringbuffer_pause), (gst_ringbuffer_stop),
- (gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
- (gst_ringbuffer_set_sample), (wait_segment),
- (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
- (gst_ringbuffer_advance), (gst_ringbuffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- Make ringbuffer faster and more simple by removing the locks
- in the playback thread.
- Add sample accurate playback based on buffer sample offsets.
- Make the baseaudiosink provide a clock.
- Parse caps in the base class.
- Correctly handle seeking, flushing and state changes.
+ ChangeLog: ChangeLog surgery.
+ Original commit message from CVS:
+ * ChangeLog:
+ ChangeLog surgery.
+ * tests/examples/seek/seek.c:
+ Move variable into ifdef too.
-2005-04-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Include config.h and check if we have X. Fixes: #540334.
+
+2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
+
+ gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
+ Original commit message from CVS:
+ Patch by: Sam Morris <sam at robots dot org to uk>
+ * gst-libs/gst/interfaces/mixertrack.c:
+ (gst_mixer_track_class_init), (gst_mixer_track_get_property),
+ (gst_mixer_track_set_property):
+ API: Add "index" property to GstMixerTrack to differantiate between
+ multiple mixer tracks with the same label.
+ * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
+ Set the "index" property of GstMixerTrack to the index given by ALSA.
+ Fixes bug #528299.
+
+2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
+ Original commit message from CVS:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/seek.c:
+ Remove libgstvideo usage. Use gtk_get_option_group instead of
+ gtk_init().
- * configure.ac:
- * gst/audioconvert/Makefile.am:
- * gst/audioscale/Makefile.am:
- Fix part of the build. Come on guys, autogen didn't even work :)
+2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Name the test registry format neutral.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Name the test registry format neutral.
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/media-info/.cvsignore:
- * gst-libs/gst/media-info/Makefile.am:
- * gst-libs/gst/media-info/README:
- * gst-libs/gst/media-info/media-info-priv.c:
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c:
- * gst-libs/gst/media-info/media-info.c:
- * gst-libs/gst/media-info/media-info.h:
- * gst-libs/gst/media-info/media-info.vcproj:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-media-info-uninstalled.pc.in:
- * pkgconfig/gstreamer-media-info.pc.in:
- Remove media-info, which is also successed by playbin (see Totem
- implementation).
+2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c:
+ Do not double notify. Remove the unsued return value.
- * configure.ac:
- * examples/Makefile.am:
- * examples/gstplay/.cvsignore:
- * examples/gstplay/Makefile.am:
- * examples/gstplay/player.c:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/play/.cvsignore:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.c:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/play/play.vcproj:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-play.pc.in:
- Remove libgstplay, playbin is now the official successor.
+2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c:
+ Also consider "speaker" as a name for master volume. If that doesn't
+ help look for the first non-mono volume control that also has a
+ playback switch.
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/xwindowlistener/Makefile.am:
- * gst-libs/gst/xwindowlistener/xwindowlistener.c:
- * gst-libs/gst/xwindowlistener/xwindowlistener.h:
- Remove deprecated xwindowlistener (I've moved xwindowlistening
- in the v4l/v4l2 plugins over to serverside).
+2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-04-25 David Schleef <ds@schleef.org>
+ ChangeLog: Forgot to save the ChangeLog :/
+ Original commit message from CVS:
+ * ChangeLog:
+ Forgot to save the ChangeLog :/
- * examples/dynparams/Makefile.am: Move demo-dparams from gst/sine
- to examples/dynparams. Examples do not belong interspersed with
- source code.
- * examples/dynparams/demo-dparams.c:
- * gst/sine/Makefile.am:
- * gst/sine/demo-dparams.c:
+2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/: Embedd the xwindow.
+ Original commit message from CVS:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/seek.c:
+ Embedd the xwindow.
+
+2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
+ (gst_ximagesink_setcaps):
+ * sys/ximage/ximagesink.h:
+ When the caps change, make sure to re-draw borders in
+ force-aspect-ratio=true mode.
+ * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
+ Don't clear the border_draw flag until we actually draw the border.
+ * tests/check/Makefile.am:
+ Ignore alsasink/src during the states test too, so it doesn't fail
+ when running without access to the sound device.
+
+2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Fix crasher when playing a parse-launch line the 2nd time.
-2005-04-25 David Schleef <ds@schleef.org>
+2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- * gst-libs/gst/xwindowlistener/Makefile.am:
+ tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c:
+ Properly ifdef tests to fix compilation.
- Convert to 0.9 API, seems to work:
- * sys/ximage/Makefile.am:
- * sys/ximage/ximagesink.c:
+2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-04-24 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ break long lines
+ Original commit message from CVS:
+ break long lines
- Link plugins against libraries:
- * ext/alsa/Makefile.am:
- * gst/tcp/Makefile.am:
+2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
- Remove asm code that should be in liboil
- * gst/videoscale/Makefile.am:
- * gst/videoscale/videoscale_x86_asm.s:
+ gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
+ Original commit message from CVS:
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gstplaybin2.c:
+ Add get-video-pad, get-audio-pad, get-text-pad action signals to
+ playbin2. This allows the user to get to the selector's sinkpads, and
+ thus inspect a range of things - caps, tags, etc.
- gettext wants these checked in:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
+2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
-2005-04-24 David Schleef <ds@schleef.org>
+ gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c:
+ Use a different constant for the convert-frame signal id.
+ Fixes #537009.
- Convert gst_main() to g_main_loop_run()
- * gst/playback/decodetest.c: (main):
- * gst/playback/test2.c: (main):
- * gst/playback/test3.c: (main):
- * gst/playback/test4.c: (main):
+2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
- Link plugins against libraries:
- * ext/libvisual/Makefile.am:
- * sys/xvimage/Makefile.am:
+ gst/playback/: Fix a whole bunch of typos in comments and log statements.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ Fix a whole bunch of typos in comments and log statements.
-2005-04-24 David Schleef <ds@schleef.org>
-
- * configure.ac: Remove idct and resample libs
- * gst-libs/gst/Makefile.am: same
-
- Remove usage of gst_library_load():
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/libvisual/visual.c: (plugin_init):
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/theora/theora.c: (plugin_init):
- * ext/vorbis/vorbis.c: (plugin_init):
- * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
- * gst/audioscale/gstaudioscale.c:
- * gst/adder/gstadder.c: (plugin_init):
- * gst/audioconvert/plugin.c: (plugin_init):
- * sys/ximage/ximagesink.c: (plugin_init):
- * sys/xvimage/xvimagesink.c: (plugin_init):
- * gst/tcp/gsttcpplugin.c: (plugin_init):
-
- Link plugins against libraries:
- * ext/ogg/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst/audioconvert/Makefile.am:
+2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
- Create proper libraries:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
+ sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Don't set colour balance values on the Xv port if the user hasn't
+ changed them (via properties or the interface). Avoids accumulating
+ rounding errors for the common case.
+ Partial fix for bug #537889.
- Move resample library to audioscale plugin directory:
- * gst-libs/gst/resample/Makefile.am:
- * gst-libs/gst/resample/README:
- * gst-libs/gst/resample/dtof.c:
- * gst-libs/gst/resample/dtos.c:
- * gst-libs/gst/resample/functable.c:
- * gst-libs/gst/resample/private.h:
- * gst-libs/gst/resample/resample.c:
- * gst-libs/gst/resample/resample.h:
- * gst-libs/gst/resample/resample.vcproj:
- * gst-libs/gst/resample/test.c:
- * gst/audioscale/Makefile.am:
- * gst/audioscale/README:
- * gst/audioscale/dtof.c:
- * gst/audioscale/dtos.c:
- * gst/audioscale/functable.c:
- * gst/audioscale/private.h:
- * gst/audioscale/resample.c:
- * gst/audioscale/resample.h:
- * gst/audioscale/test.c:
-
- Move tagedit library to gst-libs:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gsttagediting.c:
- * gst-libs/gst/tag/gsttageditingprivate.h:
- * gst-libs/gst/tag/gstvorbistag.c:
- * gst/tags/Makefile.am:
- * gst/tags/gstid3tag.c:
- * gst/tags/gstvorbistag.c:
-
- Fix for core changes:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
- (gst_sinesrc_getrange):
-
-2005-04-23 David Schleef <ds@schleef.org>
-
- * gst-libs/gst/Makefile.am: Remove idct. It hasn't been used
- in gst-plugins in a long time, and properly belongs in liboil.
- * gst-libs/gst/idct/Makefile.am:
- * gst-libs/gst/idct/README:
- * gst-libs/gst/idct/dct.h:
- * gst-libs/gst/idct/doieee:
- * gst-libs/gst/idct/fastintidct.c:
- * gst-libs/gst/idct/floatidct.c:
- * gst-libs/gst/idct/idct.c:
- * gst-libs/gst/idct/idct.h:
- * gst-libs/gst/idct/idtc.vcproj:
- * gst-libs/gst/idct/ieeetest.c:
- * gst-libs/gst/idct/intidct.c:
-
-2005-04-20 Wim Taymans <wim@fluendo.com>
+2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
- * docs/design-audiosinks.txt:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/TODO:
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_base_init),
- (gst_audiosink_class_init), (gst_audiosink_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstaudiosink.h:
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init),
- (gst_baseaudiosink_init), (gst_baseaudiosink_set_property),
- (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps),
- (gst_baseaudiosink_get_times), (gst_baseaudiosink_event),
- (gst_baseaudiosink_preroll), (gst_baseaudiosink_render),
- (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_class_init), (gst_ringbuffer_init),
- (gst_ringbuffer_dispose), (gst_ringbuffer_finalize),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked),
- (gst_ringbuffer_play), (gst_ringbuffer_pause),
- (gst_ringbuffer_resume), (gst_ringbuffer_stop),
- (gst_ringbuffer_callback), (gst_ringbuffer_delay),
- (gst_ringbuffer_played_samples), (gst_ringbuffer_commit),
- (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- An attempt at a set of audio base classes together with some
- design docs.
+ gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Ensure decodebin2 emits 'drained' signal once, and only once, when all
+ pads are drained.
-2005-04-20 Wim Taymans <wim@fluendo.com>
+2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_setcaps), (gst_audio_convert_fixate),
- (gst_audio_convert_channels):
- Link against audio libs.
- Fix audio convert plugin.
-
-2005-04-20 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter),
- (gst_ogg_demux_sink_activate):
- * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
- (theora_set_header_on_caps), (theora_enc_sink_event),
- (theora_enc_chain):
- Fix theora encoder.
-
-2005-04-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_factory_filter):
- * gst/playback/gstdecodebin.c: (find_compatibles):
- Work with staticpadtemplates in elementfactories.
-
-2005-04-12 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (compare_ranks), (print_feature), (gst_decode_bin_init),
- (dynamic_create), (dynamic_free), (find_compatibles),
- (mimetype_is_raw), (close_pad_link), (got_redirect),
- (try_to_link_1), (get_our_ghost_pad), (remove_element_chain),
- (new_pad), (no_more_pads), (unlinked), (close_link), (type_found),
- (gst_decode_bin_change_state):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (group_destroy), (group_commit),
- (check_queue), (queue_overrun), (queue_threshold_reached),
- (queue_out_of_data), (gen_preroll_element), (unknown_type),
- (new_decoded_pad), (setup_subtitle), (gen_source_element),
- (got_redirect), (setup_source), (play_base_eos),
- (gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (gst_play_bin_dispose),
- (gst_play_bin_set_property), (gen_video_element),
- (gen_text_element), (gen_audio_element), (remove_sinks),
- (gst_play_bin_send_event):
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose),
- (stream_info_change_state), (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c: (gst_stream_selector_init),
- (gst_stream_selector_get_caps), (gst_stream_selector_setcaps),
- (gst_stream_selector_request_new_pad), (gst_stream_selector_event),
- (gst_stream_selector_chain):
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (main):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps),
- (gst_xvimagesink_setcaps), (gst_xvimagesink_get_times),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_class_init):
- Raw and crude port of decodebin.
- Make playbin compile.
-
-2005-04-06 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/README:
+ apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
+ Original commit message from CVS:
+ apparently it's an error to specify nc -l -p 3000 - though the short usage
+ does not make it very clear that you can drop the host arg with -l
+
+2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
+ (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
+ Report the encoder latency. Fixes #538232.
+
+2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
+ (notify_source), (activate_group):
+ Implement the source property, emit notify when it changes in the
+ underlying uridecodebin.
+
+2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (stop_cb):
+ Free and clear the seek element list so that we don't use invalid
+ references when seeking after recreating a gst-launch line.
+
+2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_render):
+ Report latency even if we are not live instead of hiding it.
+ Take ts-offset and render-delay of the basesink into account when
+ scheduling samples.
+ Rework the clipping code so that we can take the various offsets into
+ account and still do correct clipping.
+
+2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Bump verion back to devel -> 0.10.20.1
+ Original commit message from CVS:
+ * configure.ac:
+ Bump verion back to devel -> 0.10.20.1
+
+2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
+ Don't increase the size of non-string image buffers by one as this
+ might in theory confuse decoders. Still increase it by one for string
+ image buffers to append '\0'.
+
+2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
+
+ gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
+ Fix a buffer memleak and remove a confusing and wrong debug output.
+ Fixes bug #538663.
+
+2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/appsink-src.c: Don't use a buffer after unreffing it.
+ Original commit message from CVS:
+ * examples/app/appsink-src.c: (on_new_buffer_from_source):
+ Don't use a buffer after unreffing it.
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
- (gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
- (gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
- (gst_gnomevfssrc_create), (gst_gnomevfssrc_is_seekable),
- (gst_gnomevfssrc_get_size), (gst_gnomevfssrc_start),
- (gst_gnomevfssrc_stop):
- * ext/ogg/Makefile.am:
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_get_data),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_sink_activate):
- * ext/theora/Makefile.am:
- * ext/theora/theoradec.c: (_inc_granulepos),
- (theora_dec_sink_event), (theora_dec_chain):
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_dec_chain):
- * gst-libs/gst/audio/Makefile.am:
- * sys/xvimage/Makefile.am:
- Make gnomevfssrc extend the source base class.
- Fix linking against libs in various plugins.
-
-2005-04-06 Andy Wingo <wingo@pobox.com>
-
- * gst-libs/gst/video/Makefile.am (libgstvideo_la_LDFLAGS): Use
- GST_BASE_LIBS.
-
- * configure.ac: Add check and AC_SUBST for libgstbase.
-
-2005-03-31 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/Makefile.am:
- * examples/seeking/cdparanoia.c: (main):
- * examples/seeking/cdplayer.c: (update_scale), (stop_seek),
- (play_cb), (pause_cb), (stop_cb), (main):
- * examples/seeking/playbin.c:
- * examples/seeking/seek.c: (dynamic_link), (make_mod_pipeline),
- (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
- (make_sid_pipeline), (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline),
- (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
- (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
- (make_playerbin_pipeline), (update_scale), (end_scrub), (do_seek),
- (seek_cb), (start_seek), (stop_seek), (play_cb), (pause_cb),
- (stop_cb), (main):
- * examples/seeking/spider_seek.c:
- * examples/seeking/vorbisfile.c:
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_base_init),
- (gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
- (gst_gnomevfssrc_get_property), (gst_gnomevfssrc_get),
- (gst_gnomevfssrc_open_file), (gst_gnomevfssrc_close_file),
- (gst_gnomevfssrc_getrange), (gst_gnomevfssrc_loop),
- (gst_gnomevfssrc_activate), (gst_gnomevfssrc_change_state),
- (gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
- * ext/ogg/README:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_get_type),
- (gst_ogg_pad_class_init), (gst_ogg_pad_init),
- (gst_ogg_pad_dispose), (gst_ogg_pad_finalize),
- (gst_ogg_pad_formats), (gst_ogg_pad_event_masks),
- (gst_ogg_pad_query_types), (gst_ogg_pad_getcaps),
- (gst_ogg_pad_src_convert), (gst_ogg_pad_src_query),
- (gst_ogg_pad_event), (gst_ogg_pad_reset),
- (gst_ogg_demux_factory_filter), (compare_ranks),
- (gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
- (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
- (gst_ogg_chain_new), (gst_ogg_chain_free),
- (gst_ogg_chain_new_stream), (gst_ogg_chain_get_stream),
- (gst_ogg_chain_has_stream), (gst_ogg_demux_base_init),
- (gst_ogg_demux_class_init), (gst_ogg_demux_init),
- (gst_ogg_demux_finalize), (gst_ogg_demux_handle_event),
- (gst_ogg_demux_submit_buffer), (gst_ogg_demux_seek),
- (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
- (gst_ogg_demux_get_prev_page),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
- (gst_ogg_demux_find_pad), (gst_ogg_demux_find_chain),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_chain_unlocked),
- (gst_ogg_demux_chain), (gst_ogg_demux_send_eos),
- (gst_ogg_demux_loop), (gst_ogg_demux_sink_activate),
- (gst_ogg_demux_change_state), (gst_ogg_print):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_init), (gst_ogg_mux_sinkconnect),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_push_page), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_loop):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- * ext/theora/theoradec.c: (gst_theora_dec_init), (_inc_granulepos),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event),
- (theora_dec_sink_event), (theora_dec_chain),
- (theora_dec_change_state):
- * ext/theora/theoraenc.c: (gst_theora_enc_init),
- (theora_enc_sink_setcaps), (theora_buffer_from_packet),
- (theora_push_buffer), (theora_enc_sink_event), (theora_enc_chain),
- (theora_enc_change_state):
- * ext/vorbis/Makefile.am:
- * ext/vorbis/oggvorbisenc.c:
- * ext/vorbis/oggvorbisenc.h:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_src_query), (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_dec_chain),
- (vorbis_dec_change_state):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init),
- (gst_vorbisenc_sink_setcaps), (gst_vorbisenc_init),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_chain),
- (gst_vorbisenc_change_state):
- * ext/vorbis/vorbisenc.h:
- * ext/vorbis/vorbisparse.c: (vorbis_parse_chain):
- * gst-libs/gst/audio/audioclock.c:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
- (gst_audiofilter_init), (gst_audiofilter_chain):
- * gst-libs/gst/audio/testchannels.c: (main):
- * gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
- * gst-libs/gst/media-info/media-info-priv.c: (gmip_reset),
- (gmip_find_type), (gmip_find_stream), (gmip_find_track_metadata),
- (gmip_find_track_streaminfo), (gmip_find_track_format):
- * gst-libs/gst/media-info/media-info.c:
- (gst_media_info_read_idler):
- * gst-libs/gst/play/play.c: (gst_play_get_sink_element),
- (gst_play_get_all_by_interface):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
- (gst_riff_parse_chunk), (gst_riff_parse_file_header),
- (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
- (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
- (gst_riff_parse_info):
- * gst-libs/gst/riff/riff-read.h:
- * gst-libs/gst/riff/riff.c: (plugin_init):
- * gst-libs/gst/video/Makefile.am:
- * gst-libs/gst/video/gstvideosink.c: (gst_videosink_init),
- (gst_videosink_class_init), (gst_videosink_get_type):
- * gst-libs/gst/video/videosink.h:
- * gst/audioconvert/bufferframesconvert.c:
- (buffer_frames_convert_init), (buffer_frames_convert_fixate),
- (buffer_frames_convert_setcaps), (buffer_frames_convert_chain):
- * gst/audioconvert/channelmixtest.c: (main):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_parse_caps),
- (gst_audio_convert_setcaps), (_fixate_caps_to_int),
- (gst_audio_convert_fixate), (gst_audio_convert_get_buffer),
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format),
- (gst_audio_convert_channels):
- * gst/audioconvert/gstchannelmix.h:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info), (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_configure_context), (gst_ffmpegcsp_setcaps),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_chain):
- * gst/tags/gstid3tag.c: (gst_tag_extract_id3v1_string):
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
- (mp3_type_find), (mpeg2_sys_type_find), (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (dv_type_find):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_src_negotiate),
- (gst_videotestsrc_src_link), (gst_videotestsrc_parse_caps),
- (gst_videotestsrc_src_accept_caps), (gst_videotestsrc_setcaps),
- (gst_videotestsrc_src_unlink), (gst_videotestsrc_activate),
- (gst_videotestsrc_change_state), (gst_videotestsrc_getcaps),
- (gst_videotestsrc_init), (gst_videotestsrc_src_query),
- (gst_videotestsrc_handle_src_event), (gst_videotestsrc_loop):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_fixate),
- (gst_xvimagesink_getcaps), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_change_state), (gst_xvimagesink_get_times),
- (gst_xvimagesink_show_frame), (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_expose),
- (gst_xvimagesink_set_property), (gst_xvimagesink_finalize),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- Plugin port to 0.9, ogg/theora playback should work in the seek
- example now.
- Removed old examples.
- Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
- explained in 0.9 TODO doc.
+=== release 0.10.20 ===
+
+2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * po/LINGUAS:
+ * win32/common/config.h:
+ Release 0.10.20
+ Original commit message from CVS:
+ Release 0.10.20
-2005-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * autogen.sh:
- * configure.ac:
- * ext/Makefile.am:
- * gst/Makefile.am:
- * po/POTFILES.in:
* po/af.po:
* po/az.po:
+ * po/bg.po:
+ * po/ca.po:
* po/cs.po:
+ * po/da.po:
+ * po/de.po:
* po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
* po/hu.po:
* po/it.po:
+ * po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
- * sys/Makefile.am:
- * testsuite/Makefile.am:
- remove a whole bunch of plugins. This module now contains a set
- of free reference plugins/elements as agreed.
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * examples/app/appsrc-ra.c:
+ * examples/app/appsrc-seekable.c:
+ * examples/app/appsrc-stream.c:
+ * examples/app/appsrc-stream2.c:
+ * ext/directfb/dfbvideosink.h:
+ * ext/metadata/gstbasemetadata.c:
+ * ext/metadata/gstbasemetadata.h:
+ * ext/metadata/metadata.c:
+ * ext/metadata/metadataexif.c:
+ * ext/theora/theoradec.h:
+ * gst/deinterlace2/gstdeinterlace2.h:
+ * gst/deinterlace2/tvtime/speedy.c:
+ * gst/deinterlace2/tvtime/speedy.h:
+ * gst/deinterlace2/tvtime/vfir.c:
+ Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
+ comments.
-2005-02-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
+ Original commit message from CVS:
+ 2008-06-16 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
+ (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
+ G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
+
+2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Final round of doc updates.
+ Original commit message from CVS:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/speed/gstspeed.c:
+ * gst/speexresample/gstspeexresample.c:
+ * gst/videosignal/gstvideoanalyse.c:
+ * gst/videosignal/gstvideodetect.c:
+ * gst/videosignal/gstvideomark.c:
+ * sys/dvb/gstdvbsrc.c:
+ * sys/oss4/oss4-mixer.c:
+ * sys/oss4/oss4-sink.c:
+ * sys/oss4/oss4-source.c:
+ * sys/wininet/gstwininetsrc.c:
+ Final round of doc updates.
+
+2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * docs/plugins/gst-plugins-bad-plugins.args:
+ * docs/plugins/gst-plugins-bad-plugins.hierarchy:
+ * docs/plugins/gst-plugins-bad-plugins.interfaces:
+ * docs/plugins/gst-plugins-bad-plugins.prerequisites:
+ * docs/plugins/gst-plugins-bad-plugins.signals:
+ * docs/plugins/inspect/plugin-alsaspdif.xml:
+ * docs/plugins/inspect/plugin-amrwb.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-bayer.xml:
+ * docs/plugins/inspect/plugin-bz2.xml:
+ * docs/plugins/inspect/plugin-cdaudio.xml:
+ * docs/plugins/inspect/plugin-cdxaparse.xml:
+ * docs/plugins/inspect/plugin-dtsdec.xml:
+ * docs/plugins/inspect/plugin-dvb.xml:
+ * docs/plugins/inspect/plugin-dvdspu.xml:
+ * docs/plugins/inspect/plugin-faac.xml:
+ * docs/plugins/inspect/plugin-faad.xml:
+ * docs/plugins/inspect/plugin-fbdevsink.xml:
+ * docs/plugins/inspect/plugin-festival.xml:
+ * docs/plugins/inspect/plugin-filter.xml:
+ * docs/plugins/inspect/plugin-flvdemux.xml:
+ * docs/plugins/inspect/plugin-freeze.xml:
+ * docs/plugins/inspect/plugin-gsm.xml:
+ * docs/plugins/inspect/plugin-gstinterlace.xml:
+ * docs/plugins/inspect/plugin-gstrtpmanager.xml:
+ * docs/plugins/inspect/plugin-h264parse.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-ladspa.xml:
+ * docs/plugins/inspect/plugin-metadata.xml:
+ * docs/plugins/inspect/plugin-mms.xml:
+ * docs/plugins/inspect/plugin-modplug.xml:
+ * docs/plugins/inspect/plugin-mpeg2enc.xml:
+ * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
+ * docs/plugins/inspect/plugin-mpegtsparse.xml:
+ * docs/plugins/inspect/plugin-mpegvideoparse.xml:
+ * docs/plugins/inspect/plugin-musepack.xml:
+ * docs/plugins/inspect/plugin-musicbrainz.xml:
+ * docs/plugins/inspect/plugin-mve.xml:
+ * docs/plugins/inspect/plugin-mythtv.xml
+ * docs/plugins/inspect/plugin-nas.xml:
+ * docs/plugins/inspect/plugin-neon.xml:
+ * docs/plugins/inspect/plugin-nsfdec.xml:
+ * docs/plugins/inspect/plugin-nuvdemux.xml:
+ * docs/plugins/inspect/plugin-oss4.xml
+ * docs/plugins/inspect/plugin-rawparse.xml:
+ * docs/plugins/inspect/plugin-real.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rfbsrc.xml:
+ * docs/plugins/inspect/plugin-sdl.xml:
+ * docs/plugins/inspect/plugin-sdp.xml:
+ * docs/plugins/inspect/plugin-selector.xml:
+ * docs/plugins/inspect/plugin-sndfile.xml:
+ * docs/plugins/inspect/plugin-soundtouch.xml:
+ * docs/plugins/inspect/plugin-spcdec.xml:
+ * docs/plugins/inspect/plugin-speed.xml:
+ * docs/plugins/inspect/plugin-speexresample.xml:
+ * docs/plugins/inspect/plugin-stereo.xml:
+ * docs/plugins/inspect/plugin-subenc.xml
+ * docs/plugins/inspect/plugin-timidity.xml:
+ * docs/plugins/inspect/plugin-tta.xml:
+ * docs/plugins/inspect/plugin-vcdsrc.xml:
+ * docs/plugins/inspect/plugin-videosignal.xml:
+ * docs/plugins/inspect/plugin-vmnc.xml:
+ * docs/plugins/inspect/plugin-wildmidi.xml:
+ * docs/plugins/inspect/plugin-x264.xml:
+ * docs/plugins/inspect/plugin-xvid.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * ext/amrwb/gstamrwbdec.c:
+ * ext/amrwb/gstamrwbenc.c:
+ * ext/amrwb/gstamrwbparse.c:
+ * ext/dc1394/gstdc1394.c:
+ * ext/directfb/dfbvideosink.c:
+ * ext/ivorbis/vorbisdec.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/mpeg2enc/gstmpeg2enc.cc:
+ * ext/mplex/gstmplex.cc:
+ * ext/musicbrainz/gsttrm.c:
+ * ext/mythtv/gstmythtvsrc.c:
+ * ext/theora/theoradec.c:
+ * ext/timidity/gsttimidity.c:
+ * ext/timidity/gstwildmidi.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/dvdspu/gstdvdspu.c:
+ * gst/festival/gstfestival.c:
+ * gst/freeze/gstfreeze.c:
+ * gst/interleave/deinterleave.c:
+ * gst/interleave/interleave.c:
+ * gst/modplug/gstmodplug.cc:
+ * gst/nuvdemux/gstnuvdemux.c:
+ Add missing elements to docs. Fix doc-markup: use convinience syntax
+ for examples (produces valid docbook), add several refsec2 when we
+ have several titles. Fix some types.
+
+2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
+ Original commit message from CVS:
+ * examples/app/.cvsignore:
+ * examples/app/Makefile.am:
+ * examples/app/appsink-src.c: (on_new_buffer_from_source),
+ (on_source_message), (on_sink_message), (main):
+ Add beefed up example app from bug #413418. It now also uses appsink
+ instead of fakesink for more ultimate coolness.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_init), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_unlock),
+ (gst_app_src_unlock_stop), (gst_app_src_create),
+ (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
+ (gst_app_src_end_of_stream):
+ * gst-libs/gst/app/gstappsrc.h:
+ Add block property to allow push based implementation to block when we
+ fill up the appsrc queues.
+ Emit the enough-data signal while releasing our lock.
+
+2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ examples/app/.cvsignore: Ignore more.
+ Original commit message from CVS:
+ * examples/app/.cvsignore:
+ Ignore more.
- * configure.ac:
- hunting season on 0.9 is now OPEN
+2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Do not use short_description in section docs for elements. We extract them from element details and there will be war...
+ Original commit message from CVS:
+ * ext/dc1394/gstdc1394.c:
+ * ext/ivorbis/vorbisdec.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/metadata/gstmetadatademux.c:
+ * ext/mythtv/gstmythtvsrc.c:
+ * ext/theora/theoradec.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst/bayer/gstbayer2rgb.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/rawparse/gstaudioparse.c:
+ * gst/rawparse/gstvideoparse.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpclient.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/selector/gstinputselector.c:
+ * gst/selector/gstoutputselector.c:
+ * gst/videosignal/gstvideoanalyse.c:
+ * gst/videosignal/gstvideodetect.c:
+ * gst/videosignal/gstvideomark.c:
+ * sys/oss4/oss4-mixer.c:
+ * sys/oss4/oss4-sink.c:
+ * sys/oss4/oss4-source.c:
+ Do not use short_description in section docs for elements. We extract
+ them from element details and there will be warnings if they differ.
+ Also fixing up the ChangeLog order.
+
+2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.19.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.19.3 pre-release
+
+2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fix build on win32.
+ Patch By: David Schleef <ds@schleef.org>
+ Fixes: #536874
+
+2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
+ (gst_gio_base_src_create):
+ * ext/gio/gstgiobasesrc.h:
+ Try to read the requested number of bytes, even if the first
+ read returns less than requested, until nothing is read anymore
+ or we have the requested amount of bytes. This fixes playback of
+ files via Samba as Samba only allows to read 64k at once.
+ Implement a caching algorithm that makes sure that we read at
+ least 4k of data every time. Some elements will try to read a few
+ bytes, then seek, read again a few bytes and so on and this is
+ painfully slow as every operation has to go over DBus if GVfs is
+ used as backend.
+ Fixes bug #536849 and #536848.
+ * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
+ (gst_gio_src_check_get_range):
+ Override check_get_range() to blacklist http/https URIs
+ and whitelist file URIs. More to be added on demand.
+
+2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
+ Original commit message from CVS:
+ * examples/app/Makefile.am:
+ * examples/app/appsrc-ra.c: (feed_data), (seek_data),
+ (found_source), (bus_message), (main):
+ * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
+ (found_source), (bus_message), (main):
+ * examples/app/appsrc-stream2.c: (feed_data), (found_source),
+ (bus_message), (main):
+ Added 3 more example application for using appsrc in random-access mode,
+ pull-mode streaming and pull mode seekable.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_start), (gst_app_src_do_get_size),
+ (gst_app_src_create):
+ * gst-libs/gst/app/gstappsrc.h:
+ Make stream-type property writable.
+ Unset flushing when starting so that we reuse appsrc.
+ Inform basesrc about the configured size.
+ Emit seek-data signal when we are going to a different offset in
+ random-access mode.
+
+2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
+ Original commit message from CVS:
+ * examples/app/appsrc-stream.c: (found_source), (main):
+ Use deep-notify until we can depend on a playbin2 with support for the
+ source property.
+
+2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
+ Original commit message from CVS:
+ * examples/app/.cvsignore:
+ * examples/app/Makefile.am:
+ * examples/app/appsrc-stream.c: (read_data), (start_feed),
+ (stop_feed), (found_source), (bus_message), (main):
+ Added an example on how to use appsrc in playbin in streaming mode from
+ an mmapped file.
+ * examples/app/appsrc_ex.c: (main):
+ Set pipeline to NULL to free queued buffers.
+ * gst-libs/gst/app/gstapp-marshal.list:
+ * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
+ (gst_app_src_class_init), (gst_app_src_init),
+ (gst_app_src_flush_queued), (gst_app_src_dispose),
+ (gst_app_src_set_property), (gst_app_src_get_property),
+ (gst_app_src_unlock), (gst_app_src_unlock_stop),
+ (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
+ (gst_app_src_check_get_range), (gst_app_src_do_seek),
+ (gst_app_src_create), (gst_app_src_set_stream_type),
+ (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
+ (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
+ (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
+ (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
+ (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
+ * gst-libs/gst/app/gstappsrc.h:
+ Measure max queue size in bytes instead.
+ Add support for 3 modes of operation, streaming, seekable and
+ random-access, making basesrc handle the scheduling modes for each.
+ Add appsrc:// uri handler so that automatic plugging can be done from
+ playbin2 or uridecodebin, for example.
+ Added support for custom segment formats.
+ Add support for push and pull based operations from the application.
+ Expand the methods so that errors can be detected.
+ Flush the queued buffers on seeks and when shutting down.
+ Add signals to inform the app that a seek must happen.
+
+2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.19.2 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.19.2 pre-release
+
+2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/common/: Add new API functions to the dll exports
+ Original commit message from CVS:
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgsttag.def:
+ Add new API functions to the dll exports
+
+2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ Disconnect signals from decodebins we created before we remove it from
+ playbin, to avoid crashes if the decodebin is eventually disposed after
+ the playbin itself (possible if the app takes a reference on the
+ decodebin).
+ Fixes #536521.
-2005-02-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Kick the hell out of gcc for not warning me about a symbol conflict.
+ gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
+ (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
+ (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
+ (h264_video_type_find), (mpeg_video_stream_type_find),
+ (dv_type_find), (mmsh_type_find):
+ Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
+ copy caps for no good reason (this may be desirable to make it easier
+ to detect leaks, but then it should probably be done for all caps
+ in the typefinder somewhere).
-2005-02-22 Luca Ognibene <luogni@tin.it>
+2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Do not try to run the check tests for subparse unless it has been
+ built.
+
+2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
+ Original commit message from CVS:
+ * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
+ (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
+ Do not try to run a test which requires vorbisenc unless we have
+ actually built it.
+
+2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
+ (gst_rtsp_connection_clear_auth_params),
+ (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Add a couple of missing argument guards.
+ Add a way of setting the DSCP for an RTSP connection.
+ Add an accessor method for the ip member of GstRTSPConnection as all
+ members are supposed to be private.
+
+2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
+ Fixed accidental use of IPv4 options for all IPv6 addresses.
+
+2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Document mixer track flags.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
+
+ gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
+ Don't set caps on the buffers that contain a copy of the buffer
+ including the caps of them resulting in an always increasing refcount
+ of the caps and insanely large caps. Instead include a buffer without
+ caps in the new caps. Fixes bug #536475.
+
+2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
+ Transform a given PAR to a range on the struct with the generic
+ height/width instead of the struct with the possibly restricted
+ height/width.
+
+2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
+ Prefer the given format if it contains something stricter than [1,MAX]
+ for height or width and only put a structure that requires rescaling
+ as second. This makes it possible to use videoscale in pipelines where
+ the source can actually produce the wanted height/width but usually
+ selects a different one from the requested.
+
+2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
+
+ gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
+ Original commit message from CVS:
+ Based on patch by: John Millikin <jmillikin gmail com>
+ * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
+ (gst_vorbis_tag_add_coverart):
+ Retrieve COVERART tags from vorbis comments (#512333)
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_link):
- Don't leak caps string (fixes #168134)
+2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init),
- (gst_jpegenc_init), (gst_jpegenc_finalize),
- (gst_jpegenc_change_state):
- Don't leak line buffers and context struct (fixes #168133).
+ gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
+ Don't forget to add new enum value here too (should probably use
+ glib-mkenums here...).
-2005-02-21 Tim-Philipp Müller <tim at centricular dot net>
+2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/dirac/gstdiracdec.cc:
- (gst_diracdec_chain):
- Since dirac 0.5.0 the framerate in dirac is expressed as a
- rational number. Fix build and up requirement to 0.5.0, and
- also pass parameters to gst_diracdec_link in the right order
- (fixes #167959).
+ gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
+ * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
+ * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
+ (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
+ (gst_tag_image_data_to_image_buffer):
+ Add two utility functions to avoid code duplication (#512333):
+ API: add gst_tag_image_data_to_image_buffer()
+ API: add gst_tag_list_add_id3_image()
-2005-02-21 Maciej Katafiasz <mathrick@freedesktop.org>
+2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/faad/gstfaad.c: (gst_faad_sinkconnect), (gst_faad_chain):
- * ext/faad/gstfaad.h:
- TEH LONGEST DEBUGGING SESSION EVAR is over. Fix interaction with
- certain invalid muxed streams, where some packets will contain
- junk after decoder data. Partially fixes #149158.
+ win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add gst_audio_check_channel_positions() to the exported symbols.
-2005-02-21 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain):
- Make sure we only write to writable buffers
+2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ API: Make gst_audio_check_channel_positions() public.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions):
+ * gst-libs/gst/audio/multichannel.h:
+ API: Make gst_audio_check_channel_positions() public.
+ * tests/check/libs/audio.c: (GST_START_TEST):
+ Add some simple checks for gst_audio_check_channel_positions().
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Do actually fix invalid RIFF fmt header values for alaw
- and mulaw audio instead of just saying so.
+2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Give gst_riff_create_audio_caps_with_data() a chance to
- fix up broken format header fields before extracting any
- parameters from the header. (fixes #167633)
+ sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
+ Original commit message from CVS:
+ * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
+ minrange and maxrange are scaled according to the frequency
+ multiplier.
-2005-02-19 Martin Holters <martin.holters@gmx.de>
+2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
+ Original commit message from CVS:
+ * ext/pango/Makefile.am:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
+ (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
+ Use gstvideo functions to calculate strides and plane offsets. Fixes
+ rendering issue ('ghost' images of the text on the chroma planes)
+ with widths or heights that are not multiples of 8 (#506659 and
+ probably also #485729).
+ * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
+ (main):
+ Test with odd height/width too.
- * gst/audioconvert/bufferframesconvert.c:
- (buffer_frames_convert_link):
- Don't leak othercaps. (fixes #167878)
+2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-19 Arwed v. Merkatz <v.merkatz@gmx.net>
+ gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_query_duration),
+ (gst_adder_query_latency):
+ When using gst_element_iterate_pads() one has to unref every pad
+ after usage.
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_srclink),
- (gst_visual_change_state):
- Support libvisual 0.2.0.
+2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-02-18 Tim-Philipp Müller <tim at centricular dot net>
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ Add a gtk-doc chunk for the new properties to have a Since: indication.
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_chain):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_resync), (gst_jpegenc_chain):
- Use same rowstrides for I420 as used everywhere else.
+2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-02-17 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ ChangeLog surgery, mark API change
+ Original commit message from CVS:
+ ChangeLog surgery, mark API change
+
+2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
+ (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
+ (gst_base_audio_src_change_state):
+ Provide readable actual-buffer-time and actual-latency-time properties
+ that reflect the configured ringbuffer values. Fixes #524724.
+
+2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
+ (gst_basertppayload_change_state):
+ Simply converting the running time into an RTP timestamp by scaling it
+ based on the clock-rate is good enough for making an RTP timestamp. This
+ has the added benefit that we can later on expose a property with the
+ RTP timestamp of running time 0, as is needed for RTSP servers to
+ generate the response of the PLAY request.
+
+2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (structure_has_fixed_channel_positions),
+ (gst_audio_convert_transform_caps):
+ Allow up to 11 positioned channels now that audioconvert can handle
+ this but add no default positions for > 8 channels.
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Add some unit tests for the above change: Test conversion of
+ 11 positioned channels to stereo and the other way around, test
+ conversion of 15 unpositioned channels in different ways.
+
+2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add gst_audio_clock_reset to the list of exported symbols.
+
+2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
+ Original commit message from CVS:
+ * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
+ Remove wrong_channels_identification_header unit test as we now
+ support 7 (and more channels).
+
+2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c:
+ (gst_channel_mix_fill_one_other):
+ If mixing left or right to center (or the other way around) only take
+ the complete value if we don't already have the original position in
+ the source.
+
+2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions),
+ (gst_audio_set_structure_channel_positions_list),
+ (gst_audio_fixate_channel_positions):
+ Allow rear center together with rear left/right and other previously
+ conflicting channel positions. The reason why they weren't allowed
+ was the channel mixing implementation in audioconvert.
+ Also take this into account when fixing channel layouts.
+ Allow setting channel positions for 1/2 channels when using
+ gst_audio_set_structure_channel_position().
+ * gst/audioconvert/gstchannelmix.c:
+ (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
+ (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
+ (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
+ Major rewrite of the channel mixing.
+ We now allow previously conflicting channel positions to appear
+ together (rear center and rear left/right for example).
+ Fixes bug #533817.
+ Rework the way channels are mixed together to take more possible
+ channel positions into account, properly mix from/to side channels
+ and don't assume that either center, left&right or nothing of a
+ specific position is available anymore.
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Adjust unit tests with non-standard 1/2 channel layouts to the more
+ correct new behaviour.
+ Add a unit test for 5.1->Stereo downmixing.
+
+2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
+ Add sane defaults for the 7 and 8 channel layouts as those are
+ undefined in the Vorbis spec. Use NONE channel layouts when decoding
+ more than 8 channels instead of erroring out. Fixes bug #535356.
+
+2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add theoraparse to the docs and fix some docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/theora/theoraparse.c:
+ Add theoraparse to the docs and fix some docs.
+
+2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
+ Fix EOS condition and track addition check, the track.end sector is
+ included in the track. Fixes #533265.
+
+2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
+
+ gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
+ Original commit message from CVS:
+ Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
+ (gst_video_rate_flush_prev), (gst_video_rate_event),
+ (gst_video_rate_chain):
+ * gst/videorate/gstvideorate.h:
+ React (more) to NEWSEGMENT
+ Small adjustment in timestamp calculation to prevent mismatches
+ Fixes #435633.
+
+2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/examples/seek/seek.c: Initialise error to NULL as we should.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
+ Initialise error to NULL as we should.
- * gst/avi/gstavidemux.c: (gst_avi_demux_invert):
- Declare variables at beginning of block and make gcc-2.95 happy
- (fixes # 167482, patch by Gergely Nagy).
-
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- Move some includes into the header, so that struct sockaddr_in is
- defined when it should be defined on FreeBSD as well (fixes
- #167483).
-
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
- Don't pass uninitialised values to setsockopt() here either.
+2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-17 Luca Ognibene <luogni at tin dot it>
+ gst/adder/gstadder.c: Implement latency query.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_query_duration),
+ (gst_adder_query_latency), (gst_adder_query):
+ Implement latency query.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send):
- Don't pass uninitialised values to setsockopt(). (fixes #167704)
+ gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_query_duration):
+ Correctly resync the iterator if gst_iterator_next() returns
+ GST_ITERATOR_RESYNC.
-2005-02-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybin.c: (add_sink):
- Invert bin_add/link order to workaround deadlock in opt.
+ win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
+ Original commit message from CVS:
+ * win32/vs6/libgstpbutils.dsp:
+ Add pbutils-enumtypes.c to sources (#518037).
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
+ (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
+ * gst-libs/gst/audio/gstaudioclock.h:
+ Add method to inform the clock that the time starts from 0 again. We use
+ this info to calculate a clock offset so that the time we report in
+ internal_time is monotonically increasing, as required by the clock base
+ class. Fixes #521761.
+ API: GstAudioClock::gst_audio_clock_reset()
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_change_state):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create), (gst_base_audio_src_change_state):
+ Reset reported time when we (re)create the ringbuffer.
- * gst/modplug/gstmodplug.cc:
- Add missing break causing position queries to fail.
+2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c:
+ (gst_alsa_mixer_track_update_alsa_capabilities):
+ Make sure playback volumes aren't accidentally overwritten by
+ capture volumes if an alsa mixer track has both playback and
+ capture capabilities: we create two GstMixerTracks in that
+ case, so make sure we query only the alsa capabilities that
+ refer to the type of GstMixerTrack we created from the dual
+ capability alsa element. Should fix issues with Audigy2 sound
+ cards (#518082).
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate):
- Granpos can apparently be -1, which screws up calculations...
+2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-16 Jan Schmidt <thaytan@mad.scientist.com>
+ tests/check/pipelines/oggmux.c: Don't use deprecated function.
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c: (test_pipeline):
+ Don't use deprecated function.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_send_pending_navigation),
- (gst_ximagesink_navigation_send_event), (gst_ximagesink_finalize),
- (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_send_pending_navigation),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_finalize), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h:
- Use a mutex protected list to marshal navigation
- events into the stream thread from whichever thread
- sends them.
-
-2005-02-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/speed/demo-mp3.c: (time_tick_cb), (main):
- Display current position and track length; misc. clean-ups.
-
- * gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query),
- (speed_init), (speed_chain):
- Add query function, so that the stream length and current position
- get adjusted when queried (note that current position queries may
- still be wrong if the audio sink returns values based on buffer
- timestamps instead of passing on the query).
-
-2005-02-13 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
- (gst_audio_convert_channels):
- create channel conversion matrix when linking
- * gst/audioconvert/.cvsignore:
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/channelmixtest.c: (main):
- add (ugly) test that ensures stereo <=> mono conversion works
- correctly
+2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-02-13 Benjamin Otte <otte@gnome.org>
+ gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ (gst_decode_group_control_source_pad), (gst_decode_group_expose):
+ Check for NULL cases and log them, creating ghostpads can, for example,
+ fail when the pad returns wrong caps.
+ * gst/playback/gstplaybin2.c: (perform_eos):
+ When pushing out the EOS event, collect the return value and warn when
+ something failed.
- * gst/audioconvert/gstchannelmix.h:
- include missing header file
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_compatible):
- use same sign for both channels when converting to/from compatible
- channel. Previously used different signs made the signals cancel
- each other out and appear like silence. (fixes #167269)
+2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-02-12 Tim-Philipp Müller <tim at centricular dot net>
+ gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add support for DVCPRO.
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c:
- Convert to and from YV12 (fixes #156379).
-
-2005-02-12 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_sink_link), (gst_ximagesink_change_state),
- (gst_ximagesink_chain), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_expose), (gst_ximagesink_set_property),
- (gst_ximagesink_finalize), (gst_ximagesink_init): Protect interface
- methods from chain and negotiation and vice versa (Fixes #166142).
- * sys/ximage/ximagesink.h: Add stream_lock.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose): Check for xcontext before trying to link.
-
-2005-02-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_open):
- Don't send "Hey! You gave me a NULL pointer you naughty person" as
- error message when we can't open the DVD device (when dvdnav_open()
- fails, src->dvdnav is NULL, so dvdnav_err_to_string() will return
- the above). Send something more useful instead (fixes #167117).
-
-2005-02-11 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_sink_link), (gst_xvimagesink_change_state),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose), (gst_xvimagesink_set_property),
- (gst_xvimagesink_finalize), (gst_xvimagesink_init): Protect interface
- methods from chain and negotiation and vice versa (Fixes #166142).
- Fix a possible bug of images in the buffer pool being discarded because
- we are looking at the wrong geometry.
- * sys/xvimage/xvimagesink.h: Add stream_lock.
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (crop_buffer): Change uint to
- unsigned int. (fixes #167128)
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * gst/librfb/Makefile.am: Testing stuff before committing is
- for wimps... and people with fast machines. Fix stupid
- mistake.
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * configure.ac: Pull in librfb from my CVS tree, because it is
- too small and annoying to be separate. Move rfbsrc plugin
- to gst/.
- * ext/Makefile.am:
- * ext/librfb/Makefile.am:
- * ext/librfb/gstrfbsrc.c:
- * gst/librfb/Makefile.am:
- * gst/librfb/gstrfbsrc.c:
- * gst/librfb/rfb.c:
- * gst/librfb/rfb.h:
- * gst/librfb/rfbbuffer.c:
- * gst/librfb/rfbbuffer.h:
- * gst/librfb/rfbbytestream.c:
- * gst/librfb/rfbbytestream.h:
- * gst/librfb/rfbcontext.h:
- * gst/librfb/rfbdecoder.c:
- * gst/librfb/rfbdecoder.h:
- * gst/librfb/rfbutil.h:
-
-2005-02-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/speed/Makefile.am:
- * gst/speed/demo-mp3.c: (main):
- * gst/speed/filter.func:
- * gst/speed/gstspeed.c: (speed_link), (speed_parse_caps),
- (speed_class_init), (speed_init), (speed_chain_int16),
- (speed_chain_float32), (speed_chain), (speed_set_property),
- (speed_get_property), (speed_change_state):
- * gst/speed/gstspeed.h:
- Fix speed element and make it chain-based (fixes #156467),
- and make it handle more than one channel.
-
-2005-02-10 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dts/gstdtsdec.c: (gst_dtsdec_init), (gst_dtsdec_channels),
- (gst_dtsdec_handle_event), (gst_dtsdec_handle_frame),
- (gst_dtsdec_chain), (gst_dtsdec_change_state):
- * ext/dts/gstdtsdec.h:
- Don't clobber the stack constructing the channels array.
- Make the element chain-based. DTS tracks can now be played.
-
-2005-02-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/multichannel.h:
- * gst-libs/gst/gconf/gconf.h:
- * gst-libs/gst/idct/idct.h:
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/resample/private.h:
- * gst-libs/gst/resample/resample.h:
- * gst-libs/gst/riff/riff-ids.h:
- * gst-libs/gst/video/video.h:
- * gst-libs/gst/video/videosink.h:
- Add G_BEGIN_DECLS and G_END_DECLS around headers where
- missing, so that they work when included from C++ code.
+ gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
+ Change default scaling method from nearest-neighbour to bilinear.
-2005-02-09 David Schleef <ds@schleef.org>
+2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * testsuite/gst-lint: Check for non-statically scoped
- parent_class variables. This won't be a problem once
- plugins are loaded with RTLD_LOCAL.
+ tests/check/libs/video.c: More checks.
+ Original commit message from CVS:
+ * tests/check/libs/video.c:
+ More checks.
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mplex/gstmplexibitstream.cc:
- gcc madness.
+ Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (parser_state_init),
+ (gst_sub_parse_format_autodetect), (handle_buffer):
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c: (test_tmplayer_style3b):
+ Limit duration to a maximum of five seconds for tmplayer format where
+ we can guess the duration only from the timestamp of the next line of
+ text. We don't want to show a text for eternities just because nothing
+ else is being said for a while.
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_handle_sink_event),
+ (gst_base_rtp_depayload_push_full),
+ (gst_base_rtp_depayload_change_state):
+ Check sequence numbers, mark input buffers with a discont flag for the
+ subclass when we detected a gap, drop duplicate buffers. We do this
+ because one can use the element without a jitterbuffer in front and we
+ don't want to feed the subclasses invalid or reordered data.
+ Do an error when the subclass did not provide a process function instead
+ of crashing.
+ Some other small cleanups.
+
+2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
+ May just as well use the precalculated uvstride here.
- * ext/ogg/gstogmparse.c:
- * gst/debug/gstnavigationtest.c:
- Die, thou faulty symbol pollutors (non-static parent_class).
+2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Add some documentation comments, and some new headers to be scanned.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioconvert.h:
+ * gst/gdp/gstgdpdepay.h:
+ * gst/gdp/gstgdppay.h:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcp.h:
+ Add some documentation comments, and some new headers to be scanned.
+ Rename some internal enum declarations (audioconvert's DitherType and
+ NoiseShapingType, GstUnitType from the TCP elements) to match the
+ documented GObject type names so that the docs pick them up.
+ Name the playbin2 docs markups properly so they get picked up. They'll
+ need renaming back when/if playbin2 becomes playbin.
+ 100% symbol coverage for the plugin docs, booya.
+
+2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
+ Original commit message from CVS:
+ Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
+ Fix generation of NV12/NV21 frames. Fixes bug #532454.
+
+2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * gst/playback/gstdecodebin.c: (remove_fakesink):
+ Lock the fakesink before setting the state to NULL and removing it from
+ the bin so that a concurrent state change cannot interfere.
+ Fixes #534331.
+
+2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
+
+ docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
+ Original commit message from CVS:
+ * docs/Makefile.am:
+ Fix installing plugin documentation when gtk-doc is disabled.
+
+2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
+
+ gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/Makefile.am:
+ Distribute, don't install md5.h
+
+2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
+ Original commit message from CVS:
+ 2008-05-21 Julien Moutte <julien@fluendo.com>
+ * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
+ instead of SOL_IP, works on more platforms.
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
+ arguments.
+
+2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Some debug and comment fixes.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
+ Some debug and comment fixes.
+ * tests/examples/dynamic/addstream.c: (main):
+ Fix , to ;
+
+2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Don't use bad gst_element_get_pad().
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
+ * gst/playback/decodetest.c: (new_decoded_pad_cb):
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
+ (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
+ (cleanup_decodebin):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
+ (connect_element), (gst_decode_group_control_demuxer_pad):
+ * gst/playback/gstplaybasebin.c: (queue_remove_probe),
+ (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
+ (mute_group_type):
+ * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
+ (gst_play_bin_set_property), (handoff), (gen_video_element),
+ (gen_text_element), (gen_audio_element), (gen_vis_element),
+ (remove_sinks), (add_sink), (setup_sinks):
+ * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
+ * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
+ (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
+ (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
+ (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
+ (gen_video_chain), (gen_text_chain), (gen_audio_chain),
+ (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
+ (gst_play_sink_request_pad):
+ * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
+ * gst/playback/test.c: (gen_video_element), (gen_audio_element),
+ (cb_newpad):
+ * gst/playback/test6.c: (new_decoded_pad_cb):
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ * tests/check/elements/audiorate.c: (test_injector_chain),
+ (do_perfect_stream_test):
+ * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/elements/gnomevfssink.c:
+ * tests/check/elements/textoverlay.c:
+ (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
+ * tests/check/elements/videotestsrc.c: (GST_START_TEST):
+ * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
+ * tests/check/pipelines/oggmux.c: (test_pipeline):
+ * tests/check/pipelines/streamheader.c: (GST_START_TEST):
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
+ * tests/examples/seek/scrubby.c: (make_wav_pipeline):
+ * tests/examples/seek/seek.c: (make_mod_pipeline),
+ (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
+ (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
+ (make_theora_pipeline), (make_vorbis_theora_pipeline),
+ (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
+ (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
+ (update_fill), (msg_buffering):
+ Don't use bad gst_element_get_pad().
+
+2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c:
+ Fix wrong method name in docs. Fix calculation of strf fields for
+ broken mulaw/alaw.
+ * gst-libs/gst/riff/riff-read.c:
+ Whitespace fix and removing double ';'.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/design/part-playbin2.txt: Add some leftover doc.
+ Original commit message from CVS:
+ * docs/design/part-playbin2.txt:
+ Add some leftover doc.
+
+2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
+ Fix copy & paste error in last commit.
+
+2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
+ Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
+ other channel positions when source has SIDE channels and dest doesn't
+ or the other way around.
+
+2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
+
+ gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
+ Original commit message from CVS:
+ Patch by: Henrik Eriksson <henriken at axis dot com>
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
+ (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
+ (gst_multi_fd_sink_get_property):
+ * gst/tcp/gstmultifdsink.h:
+ Add support for DSCP QOS. Fixes #469933.
+
+2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Add another test that checks if conversion between standard 1 and 2
+ channel layouts with and without positions set is working.
+
+2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions):
+ Allow non-standard 2 channel layouts.
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Add some tests for converting and remapping non-standard 1 and 2
+ channel layouts.
+
+2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c:
+ (gst_channel_mix_fill_normalize):
+ Prevent division by zero if the channel mix matrix contains only
+ zeroes.
+
+2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
+
+ gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
+ Close a buffer memory leak. Fixes bug #534071.
+
+2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ Make the GstRTSPTransport struct members public as there are no
+ setters/getters and it's supposed to be changed directly.
+ Fixes bug #533087.
+
+2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Adder also doesn't support audio/x-raw-int with width!=depth so don't
+ claim this on the pad template caps.
+
+2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_sync_latency):
+ We can only use our optimal calibration if we prerolled before the
+ latency expired.
+
+2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
+ Original commit message from CVS:
+ * configure.ac:
+ Require core CVS for GstBaseSrc buffer caps setting magic.
- * ext/mplex/gstmplexibitstream.cc:
- Fix event handling (#165525).
+2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Fix logic in last commit.
+
+2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Passthrough the channel positions if the number of output channels is
+ the same as the number of input channels, the input had a channel
+ layout and downstream requests no special one. We did this already for
+ > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
+
+2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
+ (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_received_headers_callback),
+ (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ Set the ICY caps on the srcpad from where they get picked up by the base
+ class now and set on the outgoing buffers.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
+ BaseSrc now sets the caps on outgoing buffers automatically.
+
+2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_resample_slaving),
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_change_state):
+ Change the way in which the ringbuffer is started when dealing with a
+ slaved clock and latency. We now sync to the clock until we reach
+ upstream latency before starting the ringbuffer. This has the effect
+ that we can accurately align the master and slave clocks and let the
+ rate correction code take care of the initial drift or rounding errors
+ instead of leaving them uncorrected with the old approach.
+
+2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Correctly set the default channel positions when converting to 8
+ channels.
+
+2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Error out if we don't have the required version of core.
+ Original commit message from CVS:
+ * configure.ac:
+ Error out if we don't have the required version of core.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mikmod/gstmikmod.c:
- * gst/modplug/gstmodplug.cc:
- Add missing endianness to template (fixes #165509).
+ gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
+ Use data scan helper in aac typefinder and stop scanning
+ for headers when we've found a type. Also fix potential invalid
+ memory access when calculating the frame length.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_data):
- Fix wrong order of reading of optional bytes (#165290).
+ gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
+ (mpeg_sys_is_valid_pack):
+ Don't modify scan context when we return FALSE in ensure_data, so
+ it's possible to continue scanning, and we don't end up with a NULL
+ data pointer and a positive size, which might bite us the next time
+ we're called. Small constification.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Adder doesn't support 24 bit samples so don't claim it supports them
+ in the pad template caps.
+
+2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain):
+ Validate the RTP packet before further processing it. It's just too
+ dangerous to accept random packets and people are not forced to use a
+ jitterbuffer or session manager to filter out the bad packets.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_set_extension_data),
+ (gst_rtp_buffer_get_payload_subbuffer):
+ Small cleanups.
+ When setting extension data in a buffer that is too small, we fail and
+ we should not set the extension bit.
+ Change GST_WARNINGS into g_warning because they really are
+ programming errors.
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Catch the g_warnings now in the unit tests and that fact that failing to
+ set extension data left the extension bit untouched.
+
+2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
+ Revert previous change which made basetransform handle buffer_alloc
+ and which breaks things badly in the non-passthrough case since it
+ returned buffers with a different (ie. sometimes smaller) size than
+ the size requested.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Implement FILLER event awareness.
+2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
+ Original commit message from CVS:
+ Patch by: Bernard B <b-gnome at largestprime dot net>
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
+ Fix seqnum compare function for bordercase values and fix the docs
+ again. Fixes #533075.
+ * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
+ Add a testcase for seqnum compare function.
+
+2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps),
+ (gst_adder_class_init):
+ Correctly declare the supported endianness on the pad templates
+ and check for correct endianness in the set caps function. Adder
+ only supports native endianness.
+ Also use gst_element_class_set_details_simple().
+
+2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Better debug logging in port value handling. Merging separate port
+ value loops into one.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
+
+ gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
+ Original commit message from CVS:
+ Patch by: Hannes Bistry <hannesb at gmx dot de>
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
+ * gst/tcp/gsttcpserversink.c:
+ (gst_tcp_server_sink_handle_server_read),
+ (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
+ Fix regression in clientsrc because we did not add the fd to the poll
+ set anymore. Fixes #532364.
+ Do some cleanups here and there.
+
+2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
+ Use correct marshallers. GstCaps are a boxed type and no GObject
+ subclass.
+
+2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstrtsp.def:
+ Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
+ symbols.
+
+2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * tests/check/elements/audioresample.c:
+ (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
+ (live_switch_push), (GST_START_TEST):
+ Add unit test for the latest basetransform negotiation changes.
+ See bug #526768.
+
+2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
+ Fix nv12<->nv21 conversion if stride is larger than width.
+
+2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
+
+ ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
+ Original commit message from CVS:
+ Patch by: j^ <j at oil21 dot org>
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
+ (gst_ogg_pad_parse_skeleton_fisbone):
+ * ext/ogg/gstoggdemux.h:
+ Parse presentation time from skeleton streams and use it as offset
+ for the timestamps. Fixes bug #530068.
+
+2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
+ Revert previous patch that attempted to more accurately calculate the
+ initial offset between master and slave clock. The best thing we can do
+ in general is take the time of both clocks as the diff since we don't
+ know when the actual preroll happened.
+
+2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ Fix docs: type and missing word.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_convert):
- Fix track calculations (#166208).
+2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-08 Gergely Nagy <algernon@bonehunter.rulez.org>
+ gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
+ Don't do lots of 4-byte peeks, but use the 'new' data scan helper
+ for this instead; don't check if we've found enough markers after
+ each and every step, it's enough to do that only if we've actually
+ found a new marker.
+ Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/libpng/gstpngdec.c: (gst_pngdec_init), (gst_pngdec_chain):
- * ext/libpng/gstpngenc.c:
- Fix byte-order, use proper fixed caps. Fixes #164197.
+ gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
+ (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
+ (mpeg_video_stream_type_find):
+ Move scan helper thingy to the beginning of the file so we can use
+ it in other typefind functions. Rename it to something more
+ generic. Also improve handling of things towards the end of the
+ typefind data: peek as much as we can if we know the size of the
+ data, rather than just min_size.
-2005-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/colorbalancechannel.c:
+ * gst-libs/gst/interfaces/colorbalancechannel.h:
+ * gst-libs/gst/interfaces/tuner.c:
+ * gst-libs/gst/interfaces/tunerchannel.c:
+ * gst-libs/gst/interfaces/tunerchannel.h:
+ * gst-libs/gst/interfaces/tunernorm.c:
+ * gst-libs/gst/interfaces/tunernorm.h:
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Document the GstTuner and GstColorBalance interfaces, and some
+ other random API functions that needed it. 70% symbol coverage, woo.
+
+2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
+ Choose to allocate one less segment but require one additional segment
+ as latency.
+ * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
+ No need to increment the number of segments in the source.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (clock_convert_external),
+ (gst_base_audio_sink_resample_slaving),
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_async_play):
+ Remove adding latency when returning the internal time while subtracting
+ it again when we use the value a little later.
+ When calculating the end timestamp, we are making a rounding error
+ with the current algorithm. Ensure that we don't accumulate these
+ rounding errors when aligning samples by not resampling at all if we
+ don't need to. Fixes #419351.
+ Make the initial calibration of the clock slaving a little more
+ predictable and accurate. Also handle the case where we don't do
+ clock slaving.
+
+2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
+ Original commit message from CVS:
+ Based on a patch by:
+ Björn Benderius <bjoern dot benderius at axis dot com>
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ Add conversions from/to NV12 and NV21 and conversions between those
+ two formats. Fixes bug #532166.
+
+2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
+ Abort the h264 typefinding as soon as _peek() doesn't return anything,
+ which happens for example with files smaller than 128kb.
+
+2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
+
+ gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
+ Original commit message from CVS:
+ Patch by: Wouter Cloetens <zombie at e2big dot org>
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_create), (md5_digest_to_hex_string),
+ (auth_digest_compute_hex_urp), (auth_digest_compute_response),
+ (add_auth_header), (gst_rtsp_connection_free),
+ (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
+ (gst_rtsp_connection_set_auth_param),
+ (gst_rtsp_connection_clear_auth_params):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Add Digest authorization support for RTSP connections. See #532065.
+ * gst-libs/gst/rtsp/md5.c:
+ * gst-libs/gst/rtsp/md5.h:
+ Yeap, another md5 implementation until we can depend on a glib that has
+ support for it.
+
+2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
+ Let audioresample use the buffer allocation of basetransform instead
+ of it's own stuff.
+ * tests/check/elements/audioresample.c: (alloc_only_48000),
+ (GST_START_TEST), (audioresample_suite):
+ Add unit test for the recent basetransform bugfix, where upstream
+ changes caps to something that can't be passed through anymore.
+
+2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
+ Original commit message from CVS:
+ * win32/common/config.h.in:
+ Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
+ use the real thing than having "???" unconditionally.
- * configure.ac:
- Add dvdlpcmdec
+2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_query):
+ Report the latency with the new seglatency parameter.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
+ (gst_ring_buffer_acquire):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Add new field to the ringbufferspec to specify the expected latency
+ between the underlying device read/write pointer, this is needed
+ when writing sinks that sit a little closer to the hardware.
+ Add some more docs for other fields.
+
+2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
+ Original commit message from CVS:
+ * gst-libs/gst/app/.cvsignore:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstapp-marshal.list:
+ Add marshal.list, make it compile and add to cvsignore.
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
+ (gst_app_sink_stop):
+ Small cleanups.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_init), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_unlock),
+ (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
+ (gst_app_src_create), (gst_app_src_set_caps),
+ (gst_app_src_get_caps), (gst_app_src_set_size),
+ (gst_app_src_get_size), (gst_app_src_set_seekable),
+ (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
+ (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
+ (gst_app_src_end_of_stream):
+ * gst-libs/gst/app/gstappsrc.h:
+ Beat appsrc in shape, add signals and actions.
+ Add some docs.
+ Add properties for caps, size, seekability and max-buffers.
+ Fix unlock/stop code.
+
+2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_transform_ip):
+ Return NOT_NEGOTIATED if we didn't set a process function yet for some
+ reason instead of crashing later. Might fix bug #509125.
+
+2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
+ Original commit message from CVS:
+ Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_parse_caps),
+ (structure_has_fixed_channel_positions),
+ (gst_audio_convert_transform_caps):
+ * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
+ Add support for more than 8 channels and NONE channel layouts. For
+ more than 8 channels no channel conversion is supported yet, only
+ format conversions are supported. Fixes bug #398033.
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST), (audioconvert_suite):
+ Add some unit tests by Tim for checking the NONE channel layouts
+ and more than 8 channels and add some more unit tests for channel
+ conversions.
+
+2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (connect_pad):
+ When autoplugging fails, set the element back to NULL before
+ unreffing it.
+
+2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add gst_base_audio_src_[sg]et_slave_method() to the exported
+ symbols.
+
+2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
+ Original commit message from CVS:
+ * gst/subparse/samiparse.c: (handle_start_sync),
+ (end_sami_element), (characters_sami):
+ Remove trailing, leading and double whitespaces.
+ Correctly timestamp buffers and output the last buffer too.
+ * tests/check/elements/subparse.c: (GST_START_TEST),
+ (subparse_suite):
+ Add a simple unit test for SAMI parsing.
+
+2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst/subparse/samiparse.c: (handle_start_sync),
+ (start_sami_element), (end_sami_element), (characters_sami),
+ (sami_context_reset):
+ Only output characters inside the "sync" elements. There could be
+ other elements like "style" that have some content but should
+ not be printed. Fixes bug #467911.
+
+2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.*: Start some docs.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
+ (gst_app_sink_init), (gst_app_sink_set_property),
+ (gst_app_sink_get_property), (gst_app_sink_unlock_start),
+ (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
+ (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
+ (gst_app_sink_preroll), (gst_app_sink_render),
+ (gst_app_sink_set_caps), (gst_app_sink_set_drop),
+ (gst_app_sink_get_drop):
+ * gst-libs/gst/app/gstappsink.h:
+ Start some docs.
+ Add property to drop buffers when the queue is filled
+ Fix unlocking and flushing when the queues are filled.
+
+2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (set_audio_mute),
+ (set_active_source):
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (playbin_set_audio_mute):
+ Allow setting -1 as current-audio to mute the current audio stream,
+ similar to what is done for subtitles. Fixes bug #342294.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_reset),
- (free_all_buffers), (gst_mpeg2dec_alloc_buffer):
- Don't push buffers if the src pad isn't negotiated yet.
-
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format):
- Add support for 24-bit width.
-
- * gst/dvdlpcmdec/.cvsignore:
- * gst/dvdlpcmdec/Makefile.am:
- * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_get_type),
- (gst_dvdlpcmdec_base_init), (gst_dvdlpcmdec_class_init),
- (gst_dvdlpcm_reset), (gst_dvdlpcmdec_init), (gst_dvdlpcmdec_link),
- (gst_dvdlpcmdec_chain), (gst_dvdlpcmdec_change_state),
- (plugin_init):
- * gst/dvdlpcmdec/gstdvdlpcmdec.h:
- New decoder for rearranging DVD LPCM into our audio/x-raw-int
- format. Needs support for the channels maps if someone can find
- a DVD LPCM track with > 2 channels.
-
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_handle_dvd_event),
- (gst_dvd_demux_send_discont), (gst_dvd_demux_handle_discont),
- (gst_dvd_demux_get_audio_stream), (gst_dvd_demux_process_private):
- * gst/mpegstream/gstdvddemux.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
- (gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_init_stream),
- (gst_mpeg_demux_send_subbuffer), (gst_mpeg_demux_handle_src_query):
- * gst/mpegstream/gstmpegdemux.h:
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_reset),
- (gst_mpeg_parse_parse_packhead), (gst_mpeg_parse_loop),
- (gst_mpeg_parse_get_rate), (gst_mpeg_parse_convert_src),
- (gst_mpeg_parse_handle_src_query),
- (gst_mpeg_parse_handle_src_event):
- Use audio/x-dvd-lpcm for LPCM output.
- Add DTS output.
-
-2005-02-08 Gergely Nagy <algernon@bonehunter.rulez.org>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/alpha/gstalphacolor.c: (gst_alpha_color_sink_link),
- (transform_rgb), (transform_bgr), (gst_alpha_color_chain):
- Add BGRA handling (#165736).
-
-2005-02-08 Francis Labonte <francis_labonte@hotmail.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/law/alaw-decode.c: (alawdec_link):
- * gst/law/alaw-encode.c: (alawenc_link):
- * gst/law/mulaw-decode.c: (mulawdec_link):
- * gst/law/mulaw-encode.c: (mulawenc_link):
- Fix caps memleaks (#166600).
-
-2005-02-08 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/tarkin/mem.h:
- * ext/tarkin/wavelet.h:
- * ext/tarkin/yuv.h:
- * gst/ffmpegcolorspace/avcodec.h:
- Include "_stdint.h" instead of <stdint.h>. Fixes build on
- systems that don't have stdint.h, like Solaris9 (fixes #166631).
-
-2005-02-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_clear),
- (gst_xvimagesink_change_state):
- Clear window on PAUSED->READY instead of READY->PAUSED. Stop
- Xv video (and thereby regenerate Xv colourkey) in clear() so
- that PLAY -> READY -> PLAY works (fixes #162504).
-
-2005-02-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_getcaps):
- Switch to list instead of range, since MJPEG-devices really just
- support decimations, not any size.
-
-2005-02-05 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_open_decoder),
- (gst_mpeg2dec_reset), (free_all_buffers),
- (gst_mpeg2dec_alloc_buffer), (handle_sequence):
- * ext/mpeg2dec/gstmpeg2dec.h:
- The libmpeg2 user-allocated buffer management is awkward,
- to say the least. Hopefully this fixes things.
-
-2005-02-04 Andy Wingo <wingo@pobox.com>
-
- * gst/audioconvert/bufferframesconvert.c
- (buffer_frames_convert_fixate): New function, fixates to 256
- frames per buffer by default. (Much better than 1.)
- (buffer_frames_convert_init): Set the fixate function for both src
- and sink pad.
- (buffer_frames_convert_link): After success setting nonfixed caps,
- get the negotiated caps so we can know how many buffer-frames it
- will be. No idea how this worked at all before.
-
-2005-02-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_init),
- (gst_mpeg2dec_close_decoder), (put_buffer), (check_buffer),
- (free_buffer), (free_all_buffers), (gst_mpeg2dec_alloc_buffer),
- (handle_sequence), (handle_picture):
- * ext/mpeg2dec/gstmpeg2dec.h:
- Rearrange buffer tracking and refcounting and refactor
- a little for readability.
-
-2005-02-04 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_get_type),
- (gst_v4ljpegsrc_base_init), (gst_v4ljpegsrc_class_init),
- (gst_v4ljpegsrc_init), (gst_v4ljpegsrc_src_link),
- (gst_v4ljpegsrc_getcaps), (gst_v4ljpegsrc_get):
- * sys/v4l/gstv4ljpegsrc.h:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_open), (gst_v4lsrc_src_link):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_palette_name),
- (gst_v4lsrc_get_fps):
- * sys/v4l/v4lsrc_calls.h:
- Add new v4ljpegsrc for handling the ov51x hacky "I'll give
- you jpeg inside rgb frames" driver.
- Don't error in the v4lsrc link function, just return
- REFUSED.
-
-2005-02-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
- (gst_qcamsrc_open):
- Use GST_ELEMENT_ERROR, not g_warning, if open failed.
-
-2005-02-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
- Change caps on MJPEG-B so it doesn't interfere with MJPEG/JPEG.
-
-2005-02-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_change_state):
- Reset negotiated state on PAUSED->READY.
-
-2005-02-02 David Schleef <ds@schleef.org>
-
- * configure.ac: Put DEFAULT_AUDIOSINK in config.h and use
- whereever possible. (Fixes #165997)
- * examples/capsfilter/capsfilter1.c: (main):
- * examples/dynparams/filter.c: (create_ui):
- * examples/seeking/cdparanoia.c: (get_track_info), (main):
- * examples/seeking/chained.c: (main):
- * examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
- (make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
- (make_vorbis_pipeline), (make_mp3_pipeline), (make_avi_pipeline),
- (make_mpeg_pipeline), (make_mpegnt_pipeline):
- * examples/seeking/spider_seek.c: (make_spider_pipeline):
- * examples/switch/switcher.c: (main):
- * ext/dv/demo-play.c: (main):
- * ext/faad/gstfaad.c: (gst_faad_change_state):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/smoothwave/demo-osssrc.c: (main):
- * gst-libs/gst/gconf/gconf.c: (gst_gconf_set_string),
- (gst_gconf_render_bin_from_description),
- (gst_gconf_get_default_audio_sink),
- (gst_gconf_get_default_video_sink),
- (gst_gconf_get_default_audio_src),
- (gst_gconf_get_default_video_src),
- (gst_gconf_get_default_visualization_element):
- * gst/level/demo.c: (main):
- * gst/level/plot.c: (main):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- * gst/playback/test.c: (gen_video_element), (gen_audio_element):
- * gst/playondemand/demo-mp3.c: (setup_pipeline):
- * gst/sine/demo-dparams.c: (main):
- * gst/spectrum/demo-osssrc.c: (main):
- * gst/speed/demo-mp3.c: (main):
- * gst/volume/demo.c: (main):
- * testsuite/embed/embed.c: (main):
-
-2005-02-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
- (gst_tcpclientsink_finalize):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
- (gst_tcpclientsrc_finalize):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_init), (gst_tcpserversink_finalize):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
- (gst_tcpserversrc_init), (gst_tcpserversrc_finalize):
- Don't leak the hostname when shutting down.
- In tcpserversrc, take a copy of the default hostname.
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_iso_receive):
- Set caps to systemstream=TRUE.
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * testsuite/Makefile.am:
- Fix more OSX buildbots.
-
-2005-02-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c:
- Don't send things to NULL PAD_PEERs
-
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_chain):
- Copy-on-write the incoming buffer.
-
- * gst/mpegstream/gstdvddemux.h:
- * gst/mpegstream/gstmpegclock.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (normal_seek), (gst_mpeg_demux_handle_src_event):
- * gst/mpegstream/gstmpegdemux.h:
- * gst/mpegstream/gstmpegpacketize.h:
- * gst/mpegstream/gstmpegparse.c:
- (gst_mpeg_parse_update_streaminfo), (gst_mpeg_parse_reset),
- (gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead),
- (gst_mpeg_parse_loop), (gst_mpeg_parse_get_rate),
- (gst_mpeg_parse_convert_src), (gst_mpeg_parse_handle_src_query),
- (gst_mpeg_parse_handle_src_event), (gst_mpeg_parse_change_state):
- * gst/mpegstream/gstmpegparse.h:
- * gst/mpegstream/gstrfc2250enc.h:
- Various changes to the way time is computed that make seeking and
- total time estimation much better here.
- Use G_BEGIN/END_DECLS instead of __cplusplus
-
- * gst/videocrop/gstvideocrop.c: (gst_video_crop_chain):
- Use gst_buffer_stamp instead of only copying the TIMESTAMP
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/subparse/gstsubparse.c:
- Fix OSX buildbot.
+ gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats):
+ It's SorensOn and not SorensEn.
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_enc_chain), (theora_enc_change_state):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_chain),
- (gst_vorbisenc_change_state):
- * ext/vorbis/vorbisenc.h:
- Set granulepos and timestamp correctly for streams not
- starting at 0, taking into account the initial delay.
+ gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats):
+ Fix description of video/x-flash-video.
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/mpegstream/gstdvddemux.c:
- Add audio/x-dts to audio pad template caps
+ Remove some unused code.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
+ * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
+ Remove some unused code.
+ * gst/audioconvert/gstaudioquantize.c:
+ (gst_audio_quantize_free_noise_shaping):
+ Don't return before freeing the noise shaping history.
-2005-01-30 David Schleef <ds@schleef.org>
+2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/polyp/polypsink.c: (gst_polypsink_base_init),
- (create_context), (gst_polypsink_link): Fix silly endianness
- bug. Add some debugging. Remove float from caps; it doesn't
- work. Attempt to get remote audio working.
+ tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
+ Original commit message from CVS:
+ * tests/check/elements/subparse.c: (do_test),
+ (test_tmplayer_style3b), (subparse_suite):
+ Add unit test for the tmplayer variant from bug #530962.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
- Add 3IV2 fourcc.
+ gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (handle_buffer),
+ (gst_sub_parse_sink_event):
+ * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
+ (tmplayer_parse_line):
+ Fix parsing of tmplayer subtitle variant where every single line contains
+ text and there isn't an empty line after each line to determine the
+ duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
+ making sure that we push out the last line of text without a duration if
+ there's still text left in the buffer at the end.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream), (swap_line),
- (gst_avi_demux_invert), (gst_avi_demux_process_next_entry),
- (gst_avi_demux_stream_data):
- * gst/avi/gstavidemux.h:
- Invert DIB images. Fixes #132341.
+ gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (feed_textbuf):
+ Fix detection of discontinuities based on the buffer offset (doesn't work
+ so well if no buffer offset is set) and also check for the DISCONT buffer
+ flag. This keeps the parser state from being reset after each buffer in
+ the unit test.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- D'oh, reference the palette data, not the palette structure.
- Fixes color distortion in #132341.
+ gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
+ Further fine-tuning: don't absolutely require sequence or GOP headers
+ (as introduced in the previous commit), but adjust the typefind
+ probabilities returned accordingly if we don't see them. Also make sure
+ picture header and first slice are somewhat close to each other (which
+ is not perfect but still better than requiring a fixed offset or having
+ no limit at all).
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
+ (gst_basertppayload_sink_setcaps),
+ (gst_basertppayload_sink_getcaps):
+ Rename the setcaps/getcaps function internally to make it clear that
+ they are called for the sink pad.
+
+2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init),
+ (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
+ (gst_base_rtp_depayload_packet_lost),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Catch packet-lost events from the jitterbuffer and convert them into a
+ vmethod call (lost-packet) so that depayloaders can do something smart.
+ Also add a default packet-lost function that sends out a segment update
+ to the decoders.
+
+2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
+ Original commit message from CVS:
+ * gst/playback/test4.c:
+ * gst/playback/test5.c:
+ * gst/playback/test6.c:
+ * gst/playback/test7.c:
+ Also include config.h when relying on defines from it. Fixes the
+ build. Its been a please to serve :)
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- PAR can be non-fixed when not provided as argument (#162626).
+2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
-2005-01-29 David Moore <dcm@acm.org>
+ * ChangeLog:
+ * gst/videotestsrc/videotestsrc.c:
+ Add support for NV12 and NV21 in videotestsrc
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
+ (paint_setup_NV21), (paint_hline_NV12_NV21):
+ Add support for NV12 and NV21 in videotestsrc
+
+2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
+ * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
+ (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
+ (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
+ (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
+ (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
+ (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
+ (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
+ (vs_image_scale_linear_RGB555):
+ Support 1x1 images as input and output as for example the BBC HQ new
+ streams have 1x1 GIFs in the playlists for some reason.
+
+2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
+ (try_to_link_1):
+ If we can't activate one of the decoders we plugged in (such as,
+ say, musepackdec) for some reason (it might not support push mode,
+ for example), remove any pad probes that close_pad_link() might
+ have set up. This makes sure we later don't try to remove a probe
+ for a pad that doesn't exist any longer, and avoids nast warnings
+ and probably other things too.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header):
- Re-apply patch from #142272 that allows non-seekable sources,
- re-proposed by Daniel Drake <dsd@gentoo.org>.
+ gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
+ (plugin_init):
+ Rework mpeg video stream typefinding a bit more: make sure sequence,
+ GOP, picture and slice headers appear in the order they should and
+ that we've in fact at least had one of each; fix picture header
+ detection; decouple picture and slice header check - don't assume
+ they're at a fixed offset, there may be extra data in between. Also,
+ announce varying degrees of probability depending on what we found
+ exactly (multiple pictures, at least one picture, just sequence and
+ GOP headers). Finally, in _ensure_data(), take into account that we
+ might be typefinding smaller amounts of data, such as the first
+ buffer of a stream, so fall back to the minimum size needed as long
+ as that's available, instead of erroring out if there's less than
+ 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
+ fuzzed file from #399342 as valid.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
- * gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_init):
- Use the src template for creating the src pad (#162330).
+ ext/theora/theoradec.c: Cool kids don't divide by zero.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c:
+ Cool kids don't divide by zero.
+ Treat PAR of x:0 as 1:1.
+ Fixes #530719.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/musepack/Makefile.am:
- * ext/musepack/gstmusepackdec.c: (gst_musepackdec_class_init),
- (gst_musepackdec_init), (gst_musepackdec_dispose),
- (gst_musepackdec_src_query), (gst_musepackdec_src_convert),
- (gst_musepack_stream_init), (gst_musepackdec_loop),
- (gst_musepackdec_change_state):
- * ext/musepack/gstmusepackdec.cpp:
- * ext/musepack/gstmusepackdec.h:
- * ext/musepack/gstmusepackreader.c: (gst_musepack_reader_peek),
- (gst_musepack_reader_read), (gst_musepack_reader_seek),
- (gst_musepack_reader_tell), (gst_musepack_reader_get_size),
- (gst_musepack_reader_canseek), (gst_musepack_init_reader):
- * ext/musepack/gstmusepackreader.cpp:
- * ext/musepack/gstmusepackreader.h:
- Update to 1.1 API (#165446).
-
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
+ (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
+ (mpeg_video_stream_type_find):
+ Refactor a bit: use context structure to track parsing offset and size of
+ available data and make the code a bit clearer. Fixes bad memory access
+ in #356937.
+
+2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
+ Original commit message from CVS:
+ * gst/playback/test4.c:
+ * gst/playback/test5.c:
+ * gst/playback/test6.c:
+ * gst/tcp/gstmultifdsink.c:
+ Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
+ is defined.
- * ext/Makefile.am:
- Unbreak buildbot.
+2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Clarify some docs.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
+ (gst_base_audio_src_class_init), (gst_base_audio_src_init),
+ (gst_base_audio_src_set_slave_method),
+ (gst_base_audio_src_get_slave_method),
+ (gst_base_audio_src_set_property),
+ (gst_base_audio_src_get_property), (gst_base_audio_src_create):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ Add property and methods for selecting the clock slave method in the
+ source, like in the sink.
+ We only implement "none" and "re-timestamp" for now.
+ API: gst_base_audio_src_set_slave_method()
+ API: gst_base_audio_src_get_slave_method()
+
+2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.*: Add more docs.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
+ (gst_app_sink_init), (gst_app_sink_set_property),
+ (gst_app_sink_get_property), (gst_app_sink_event),
+ (gst_app_sink_preroll), (gst_app_sink_render),
+ (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
+ (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
+ (gst_app_sink_pull_buffer):
+ * gst-libs/gst/app/gstappsink.h:
+ Add more docs.
+ Add signals for when preroll and render buffers are available.
+ Add property to control signal emission.
+ Add property to control the max queue size.
+
+2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ Fix the docs about the seqnum compare function, it returns a difference.
+
+2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
+ Original commit message from CVS:
+ * ext/alsa/gstalsadeviceprobe.c:
+ (gst_alsa_get_device_list): Don't return before freeing up
+ the allocated structures.
+
+2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ Remove obsolete streaminfo code and fix a leak. Fixes #529546
-2005-01-28 Andy Wingo <wingo@pobox.com>
+2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/dv/gstdvdec.c: Change the pixel aspect ratio of dvdec output
- to reflect a different dubious internet source. Add a reference
- and some commentary.
+ ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c:
+ Revert the event part, that should not go in.
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gststreamselector.c: (gst_stream_selector_init),
- (gst_stream_selector_get_caps), (gst_stream_selector_chain):
- * gst/playback/gststreamselector.h:
- Be more selective when we're redoing caps negotiation from
- within the chain function on a stream change.
+ ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c:
+ Don't leak GstPluginFeatures when filtering.
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- * ext/Makefile.am:
- * ext/amrnb/Makefile.am:
- * ext/amrnb/amrnb.c: (plugin_init):
- * ext/amrnb/amrnbdec.c: (gst_amrnbdec_get_type),
- (gst_amrnbdec_base_init), (gst_amrnbdec_class_init),
- (gst_amrnbdec_init), (gst_amrnbdec_link), (gst_amrnbdec_chain),
- (gst_amrnbdec_state_change):
- * ext/amrnb/amrnbdec.h:
- * ext/amrnb/amrnbparse.c: (gst_amrnbparse_get_type),
- (gst_amrnbparse_base_init), (gst_amrnbparse_class_init),
- (gst_amrnbparse_init), (gst_amrnbparse_formats),
- (gst_amrnbparse_querytypes), (gst_amrnbparse_query),
- (gst_amrnbparse_handle_event), (gst_amrnbparse_reserve),
- (gst_amrnbparse_loop), (gst_amrnbparse_state_change):
- * ext/amrnb/amrnbparse.h:
- Add support for AMR-NB (mobile phone audio format; #155163, #163286).
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add AMR-NB/-WB raw formats.
- * ext/alsa/gstalsa.c: (gst_alsa_link):
- Keep valid time when changing format.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse_trak):
- Add some more format-specific options (#140141, #143555, #155163).
-
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix logic error in timing of subtitle stream synchronization.
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- Add skip-chunk, which is found in kodak-camera streams.
-
-2005-01-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Add some logging for cases when grabbing the xv failed.
- * po/LINGUAS:
- * po/vi.po:
- Adding Vietnamese translation (submitted by Clytie Siddall)
-
-2005-01-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Use realpad for signal.
-
-2005-01-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstid3demuxbin.c: (gst_id3demux_bin_base_init):
- Fix category so decodebin picks it up.
-
-2005-01-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/Makefile.am:
- * ext/mad/gstid3demuxbin.c: (gst_id3demux_bin_get_type),
- (gst_id3demux_bin_base_init), (gst_id3demux_bin_class_init),
- (gst_id3demux_bin_init), (gst_id3demux_bin_remove_pad),
- (found_type), (gst_id3demux_bin_change_state):
- * ext/mad/gstid3tag.c: (gst_id3_tag_add_src_pad),
- (gst_id3_tag_init), (gst_id3_tag_handle_event),
- (gst_id3_tag_src_link), (gst_id3_tag_chain),
- (gst_id3_tag_change_state), (plugin_init):
- * ext/mad/gstmad.h:
- Add id3demuxbin (which is a simple bin consisting of id3demux
- and typefind), take over rank from id3demux, remove typefind
- code from id3demux. Makes all broken mp3s that I know of work,
- and thereby fixes #152688.
-
-2005-01-27 Edward Hervey <bilboed@bilboed.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstmad.c: (gst_mad_src_event):
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
- Allow seeks on audio pad, make mad forward those (#164826).
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Set duration (#165335).
-
-2005-01-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
- (gst_asf_demux_commit_taglist), (gst_asf_demux_process_comment),
- (gst_asf_demux_process_ext_content_desc),
- (gst_asf_demux_change_state), (gst_asf_demux_add_audio_stream),
- (gst_asf_demux_add_video_stream), (gst_asf_demux_setup_pad):
- * gst/asfdemux/gstasfdemux.h:
- Improve metadata display, e.g. if the metadata comes before the
- streams are loaded (which is perfectly valid).
-
-2005-01-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * tools/gst-launch-ext-m.m:
- Fix AVI/ASF pipelines (#165340).
-
-2005-01-26 Amaury Jacquot <sxpert@esitcom.org>
- * ext/cairo/gsttextoverlay.c: include string.h and strings.h to fix
- build failure on amd64
-
-2005-01-26 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/mad/gstid3tag.c: (mad_id3_parse_latin1_string),
- (mad_id3_parse_comment_frame), (gst_mad_id3_to_tag_list):
- Check environment variables GST_ID3V2_TAG_ENCODING,
- GST_ID3_TAG_ENCODING and GST_TAG_ENCODING for a colon-separated
- list of character encodings to force interpretation of non-unicode
- strings stored in an ID3v2 tag to a particular encoding. If none
- is specified, try to use current locale's encoding, then fall back
- to ISO-8859-1 (which will always succeed). (Resolves #149274)
- * gst/tags/gstid3tag.c: (gst_tag_from_id3_tag),
- (gst_tag_extract_id3v1_string), (gst_tag_list_new_from_id3v1):
- Check environment variables GST_ID3V1_TAG_ENCODING,
- GST_ID3_TAG_ENCODING and GST_TAG_ENCODING for a colon-separated
- list of character encodings to use in case a string encountered
- in an ID3v1 tag is not valid UTF-8 already. If no encoding is
- specified, try to use the current locale's encoding, then fall
- back to ISO-8859-1 (which will always succeed).
-
-2005-01-25 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
- - on half framerate, compute the rate in advance so the comparisons
- don't compare wrong values
- - don't use mad_synth/frame_mute anymore, this mirrors mad_decoder
- behaviour
- - don't use mad_header_decode anymore, mad_frame_decode does that
- automatically
- - when getting rid of consumed bytes, reset the stream's skiplen
- (fixes #163867)
-
-2005-01-26 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init)
- Use 1/2 a second for default max_discont, as PES streams from DVB
- seem to have larger spacings in the SCR.
- Fix a typo.
-
-2005-01-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c: (group_commit):
- Notify delayed stream-info availability.
-
-2005-01-26 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/a52dec/gsta52dec.c: (gst_a52dec_push),
- (gst_a52dec_handle_event), (gst_a52dec_chain):
- Add some debug output. Check that a discont has a valid
- time associated.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
- (gst_alsa_sink_loop):
- Ignore TAG events. A little extra debug for broken timestamps.
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init), (dvdnavsrc_loop),
- (dvdnavsrc_change_state):
- Ensure we send a discont to engage the link before we send any
- other events.
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_init),
- (dvdreadsrc_finalize), (_close), (_open), (_seek_title),
- (_seek_chapter), (seek_sector), (dvdreadsrc_get),
- (dvdreadsrc_uri_get_uri), (dvdreadsrc_uri_set_uri):
- Handle URI of the form dvd://title[,chapter[,angle]]. Currently only
- dvd://title works in totem because typefinding sends a seek that ends
- up going back to chapter 1 regardless.
- * ext/mpeg2dec/gstmpeg2dec.c:
- * ext/mpeg2dec/gstmpeg2dec.h:
- Output correct timestamps and handle disconts.
- * ext/ogg/gstoggdemux.c: (get_relative):
- Small guard against a null dereference.
- * ext/pango/gsttextoverlay.c: (gst_textoverlay_finalize),
- (gst_textoverlay_set_property):
- Free memory when done. Don't call gst_event_filler_get_duration on
- EOS events. Use GST_LOG and GST_WARNING instead of g_message and
- g_warning.
- * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init),
- (draw_line), (gst_smoothwave_dispose), (gst_sw_sinklink),
- (gst_sw_srclink), (gst_smoothwave_chain):
- Draw solid lines, prettier colours.
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
- Add a default palette that'll work for some movies.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init),
- (gst_dvd_demux_handle_dvd_event), (gst_dvd_demux_send_discont),
- (gst_dvd_demux_send_subbuffer), (gst_dvd_demux_reset):
- * gst/mpegstream/gstdvddemux.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
- (gst_mpeg_demux_parse_syshead), (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
- (gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead):
- * gst/mpegstream/gstmpegparse.h:
- Use PTM/NAV events when for timestamp adjustment when connected to
- dvdnavsrc. Don't use many discont events where one suffices.
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gen_preroll_element), (gst_play_base_bin_add_element):
- * gst/playback/gstplaybasebin.h:
- Make sure we remove subtitles from the same bin we put them in.
- * gst/subparse/gstsubparse.c: (convert_encoding), (parse_subrip),
- (gst_subparse_buffer_format_autodetect),
- (gst_subparse_change_state):
- Fix some memleaks and invalid accesses.
- * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find),
- (oggskel_type_find), (cmml_type_find), (plugin_init):
- Some typefind functions for Annodex v3.0 files
- * gst/wavparse/gstwavparse.h:
- GstRiffReadClass is the correct parent class.
-
-2005-01-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add extradata to huffyuv (fixes #165013).
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data):
- Fix extradata extraction if it is in the chunk size.
+ ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
+ packet. Should conform to what we currently think is the
+ final Ogg/Dirac muxing spec.
+
+2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
+
+ sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Fix typo that causes the overlay keying color to bright green
+ on a 16-bit display. Dark grey good. Bright green bad.
-2005-01-25 Edward Hervey <bilboed@bilboed.com>
+2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfsuri.c:
+ Add FIXME comment about using uri-list for source and sink.
- * gst/effectv/gstquark.c: (gst_quarktv_class_init),
- (gst_quarktv_change_state), (gst_quarktv_dispose):
- Memory free'ing location fix (#164708).
+2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
+ GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
+ vaargs functions to gint. Otherwise the fractions will get 0 set
+ instead of the correct value on big endian systems. Fixes bug #529018.
+
+2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ (gst_gnome_vfs_sink_uri_get_protocols):
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ (gst_gnome_vfs_src_uri_get_protocols):
+ * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
+ (gst_gnomevfs_get_supported_uris):
+ Get the list of supported URI schemes in a threadsafe way and use the
+ same list for the source and sink.
+
+2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (_internal_get_supported_protocols),
+ (gst_gio_get_supported_protocols):
+ Don't generate a new supported protocols list on each call but cache
+ it. It's supposed to be static anyway, this way we only leak it once
+ per process.
+ * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
+ (gst_gio_sink_class_init), (gst_gio_sink_finalize),
+ (gst_gio_sink_set_property), (gst_gio_sink_get_property),
+ (gst_gio_sink_start):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
+ (gst_gio_src_class_init), (gst_gio_src_finalize),
+ (gst_gio_src_set_property), (gst_gio_src_get_property),
+ (gst_gio_src_start):
+ * ext/gio/gstgiosrc.h:
+ API: Add "file" properties where one can set a GFile as source/destination.
+ Add locking to the properties and use gst_element_class_set_details_simple()
+ instead of a static GstElementDetails struct.
+
+2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
+ (plugin_init):
+ Add "mpp" and "mp+" as possible extensions for MusePack files.
+ Add typefinding for MusePack StreamVersion 8 files and include the
+ stream version in the caps.
-2005-01-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/playback/gstplaybasebin.c: (group_commit),
- (gen_preroll_element), (probe_triggered), (gen_source_element),
- (setup_source), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element):
- Don't block for streams.
- * gst/playback/gststreaminfo.c: (stream_info_change_state),
- (gst_stream_info_set_mute):
- Use gst_pad_set_active_recursive.
+ gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ (gst_rtp_payload_info_for_name):
+ Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
-2005-01-25 Andy Wingo <wingo@pobox.com>
+2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lelement.c (gst_v4l_iface_supported): Fix compile
- for #ifndef HAVE_XVIDEO.
+ configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
+ Original commit message from CVS:
+ * configure.ac:
+ Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
+ (NB: this only affects compilation of some of the examples).
+ Remove some configure.ac cruft that's not needed any longer.
-2005-01-24 Jeffrey C. Ollie
+2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
- reviewed by: Maciej Katafiasz <mathrick@freedesktop.org>
+ gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ Don't validate the payload if there isn't any.
+ Fixes #525915
- * ext/gsm/gstgsmdec.c: (gst_gsmdec_init), (gst_gsmdec_chain):
- * ext/gsm/gstgsmdec.h:
- * ext/gsm/gstgsmenc.c: (gst_gsmenc_init), (gst_gsmenc_chain):
- * ext/gsm/gstgsmenc.h:
- Fix rate to 8kHz as per spec, removes obscure errors when no rate
- was given by property. Add proper buffer timestamps and offsets.
+2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
+ Use g_atomic_int_set() instead of gst_atomic_int_set().
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Audio can be <8000Hz.
+2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Return NULL instead of a gchar * array with one NULL element if we
+ don't get any supported URI schemes from GIO.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Explicit state change to workaround refcount bugs.
+2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ Remove cpp style commented old code.
- * gst/avi/gstavimux.c: (gst_avimux_write_tag),
- (gst_avimux_riff_get_avi_header):
- Fix...
+2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstdecodebin2.c: Fix signal docs.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Fix signal docs.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
- (gst_riff_read_element_data):
- * gst-libs/gst/riff/riff-read.h:
- Add _peek version (req'ed in CDXA).
- * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
- (gst_cdxaparse_loop):
- Fix parsing in playbin.
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Ignore current_ pads, they cause major annoyance.
+2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
+ (gst_text_overlay_init):
+ Fix textoverlay unit test again by making the supposed default
+ value for the wait-text property the actual default value.
+ Also fix Since: tag for new property.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Safety guard.
+2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
+ (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
+ (gst_video_format_get_pixel_stride),
+ (gst_video_format_get_component_width),
+ (gst_video_format_get_component_height),
+ (gst_video_format_get_component_offset), (gst_video_format_get_size),
+ (gst_video_format_convert):
+ Add guards to these functions to ensure sane input values.
+ * tests/check/libs/video.c:
+ Fix unit test not to create caps with width=0 and height=0.
- * gst/avi/gstavimux.c: (gst_avimux_write_tag):
- Fix padding...
+2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/design/draft-keyframe-force.txt: Fix typo.
+ Original commit message from CVS:
+ * docs/design/draft-keyframe-force.txt:
+ Fix typo.
+ * gst/playback/gstqueue2.c: (update_buffering),
+ (gst_queue_handle_src_query):
+ Set buffering mode in the messages.
+ Set buffering percent in the query.
+ * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
+ (do_stream_buffering), (do_download_buffering), (msg_buffering):
+ Do some more fancy things based on the buffering method in use.
+
+2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
+ (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
+ (msg_buffering), (main):
+ Add basic download reports to seek using the new buffering API.
+
+2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (update_buffering),
+ (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
+ (gst_queue_src_checkgetrange_function):
+ Include extra buffering stats in the buffering message.
+ Implement BUFFERING query.
+ * gst/playback/gsturidecodebin.c: (do_async_start),
+ (do_async_done), (type_found), (setup_streaming), (setup_source),
+ (gst_uri_decode_bin_change_state):
+ Only add decodebin2 when the type is found in streaming mode.
+ Make uridecodebin async to PAUSED even when we don't have decodebin2
+ added yet.
+
+2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Filter cdda from the supported URI schemes. We can't support
+ musicbrainz tags and everything else one expects from a cdda source
+ with GIO. Fixes bug #526794.
+
+2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * sys/xvimage/xvimagesink.c:
+ Fix calculation of 'expected size' for YV12 buffers.
+ Original commit message from CVS:
+ 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
+ * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_buffer_alloc):
+ Fix calculation of 'expected size' for YV12 buffers.
+ Be a little more verbose in the debug output for buffer-alloc'ed
+ buffers which turn out to have the wrong size.
- * gst/matroska/ebml-read.c: (gst_ebml_read_buffer):
- Allow for 0-sized buffers. Fixes length query problems in
- starwars.mkv from the testsuite.
+2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ Fix calculation of 'expected size' for YV12 buffers.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_buffer_alloc):
+ Fix calculation of 'expected size' for YV12 buffers.
+ Be a little more verbose in the debug output for buffer-alloc'ed
+ buffers which turn out to have the wrong size.
- * gst/videobox/gstvideobox.c: (gst_video_box_copy_plane_i420),
- (gst_video_box_i420), (gst_video_box_chain):
- Fix row strides for I420 (fixes #163159)
-
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
- MPEG2 has a useful rate property, so we can actually use that.
- For MPEG-1, continue using the bytes/time properties.
+ Merge other changes from 0.10.19 release branch.
+ Original commit message from CVS:
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-base.doap:
+ Merge other changes from 0.10.19 release branch.
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add intel-h263.
+ gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ * gst/playback/gstplayback.c: (plugin_init):
+ * gst/volume/gstvolume.c: (plugin_init):
+ Work around missing bits of thread-safety on older GLibs some
+ more to avoid assertions when starting up multiple playbin
+ objects concurrently (see #512382).
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
- Fail if caps negotiation fails. Should fix #162184, and should
- definately be in there regardless of it fixing the actual bug.
- * gst/avi/gstavimux.c: (gst_avimux_get_type), (gst_avimux_init),
- (gst_avimux_write_tag), (gst_avimux_riff_get_avi_header),
- (gst_avimux_riff_get_avix_header),
- (gst_avimux_riff_get_video_header),
- (gst_avimux_riff_get_audio_header), (gst_avimux_write_index),
- (gst_avimux_start_file), (gst_avimux_handle_event),
- (gst_avimux_change_state):
- * gst/avi/gstavimux.h:
- Refactor structure writing to use GST_WRITE_UINT macros, add
- metadata writing support.
+ gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
+ Remove some more fields.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
- Elements may already be destroyed when this function is called.
+ configure.ac: Actually build dlls when cross-compiling with mingw32.
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * configure.ac:
+ Actually build dlls when cross-compiling with mingw32.
+ Fixes bug #526247.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (gst_qtdemux_handle_esds):
- More memory leak fixes (#149162).
+ configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_add_stream):
- Fix two memleaks.
+ tests/examples/seek/seek.c: Add statusbar.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
+ (msg_buffering), (connect_bus_signals), (main):
+ Add statusbar.
+ Add buffering support with feedback in the statusbar.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
- Argh...
+ ext/ogg/gstoggmux.c: Fix sample pipeline description.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Fix sample pipeline description.
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
- Fix off-by-one bug. Fixes warnings during playback of sincity.mp4
- when fixating to six channels in Totem.
+ docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Update introspection data.
+ * ext/ogg/gstoggmux.c:
+ Document oggmux.
+ * gst/playback/gstdecodebin2.c:
+ Don't use gtk-doc style comment start for private stuff, but make it
+ formatted like this for consistency.
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_init), (gst_decode_bin_dispose),
+ (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
+ (gst_decode_bin_set_property), (gst_decode_bin_get_property),
+ (analyze_new_pad), (connect_pad), (expose_pad),
+ (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
+ (gst_decode_group_expose), (gst_decode_group_free),
+ (do_async_start), (do_async_done), (gst_decode_bin_change_state):
+ Remove fakesink hack, we can now implement this more elegantly.
+ Added property to bypass typefinding.
+ Removed underrun callback and demuxer pad probe, we now use the srcpad
+ probe to expose groups.
+ API::sink-caps property
+ * gst/playback/gstplaybin2.c: (no_more_pads_cb):
+ Guard against multiple emissions of the no_more_pads signal, which
+ happens when we are dealing with chained oggs.
+ * gst/playback/gsturidecodebin.c: (remove_decoders),
+ (make_decoder), (type_found), (setup_streaming), (source_new_pad),
+ (setup_source):
+ For streams, use our own typefind element and plug our queue after it.
+ We will need this to determine the type of buffering to use for the
+ queue soon.
+
+2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
+ Guard against over and underflows because of clock slaving.
+ When we are using our own clock, still compensate for any calibrations
+ that we might have done to our clock.
+
+2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_type_packet),
+ (theora_dec_chain):
+ Don't try to do anything fancy with the return code from pushing an
+ event, it does not have enough information to turn it into a
+ GST_FLOW_ERROR.
+
+2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Add small debug line.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
+ (gst_ogg_demux_chain_elem_pad):
+ Add small debug line.
+ Pass return code from the internal decoder instead of the too generic
+ GST_FLOW_ERROR.
+
+2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/base64.c:
+ * gst-libs/gst/cdda/base64.h:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cddabasesrc_calculate_musicbrainz_discid):
+ Use GLib's base64 implementation instead of our own.
+
+2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
+ (gst_ogg_demux_read_chain):
+ Refix oggdemux, we only have a problem if we failed to find a chain and
+ we are not EOF.
+
+2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
+
+ ext/ogg/gstoggdemux.c: When we fail to find a BOS page and we and up with no chain, error out properly instead of seg...
+ Original commit message from CVS:
+ Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
+ (gst_ogg_demux_read_chain):
+ When we fail to find a BOS page and we and up with no chain, error out
+ properly instead of segfaulting. Fixes #525665.
+
+2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
+ The new-pad-group sequence is add-pads, no-more-pads, add-pads,
+ no-more-pads...
+
+2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (update_out_rates),
+ (gst_queue_open_temp_location_file),
+ (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
+ (gst_queue_handle_src_query), (gst_queue_set_property):
+ Update the estimated input data when we push out a buffer.
+ Add some debug info about the temp file.
+ Only forward src events when we are not using a temp file.
+ Don't block the duration query, we need to find something better.
+ Don't leak the temp filename.
+
+2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: Require GLib 2.12 and liboil 0.3.14.
+ Original commit message from CVS:
+ * configure.ac:
+ Require GLib 2.12 and liboil 0.3.14.
+ * gst/volume/gstvolume.c: (volume_process_double):
+ Unconditionally use liboil 0.3.14 function.
+
+2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ ms-gsm can have arbitrarty sample rates. See #481354.
+
+2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ MP4S is generic MPEG-4, not a microsoft variant.
+
+2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ Check the body CRC (if set) when depayloading.
+ Fixes #522401.
- * ext/dvdread/dvdreadsrc.c: (get_next_cell_for):
- Fix compile warnings on Solaris 10 buildbot
+2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+ ext/pango/gsttextoverlay.c: Fix Since: version for new property.
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
+ Fix Since: version for new property.
- * ext/dvdread/dvdreadsrc.c: (_read):
- Don't read beyond the last cell in a chapter (fixes
- invalid memory access)
+2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
+ (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
+ Don't error when poll_wait returns EAGAIN.
+
+2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_is_filled):
+ The queue is never filled when there are no buffers in the queue at all.
+ Fixes #523993.
+
+2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Update some docs.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (init_group), (free_group), (gst_play_bin_init),
+ (gst_play_bin_finalize), (gst_play_bin_set_uri),
+ (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
+ (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
+ (gst_play_bin_set_current_video_stream),
+ (gst_play_bin_set_current_audio_stream),
+ (gst_play_bin_set_current_text_stream),
+ (gst_play_bin_set_encoding), (gst_play_bin_set_property),
+ (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
+ (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
+ (activate_group), (deactivate_group), (setup_next_source),
+ (save_current_group), (gst_play_bin_change_state):
+ Update some docs.
+ Add new locks and conds to protect pipeline creation and group
+ switching.
+ Implement the sub-uri property.
+ Keep track of pending uridecodebin creation and configure the output
+ pipeline after all streams are configured.
+ Propagate subtitle encoding to the uridecodebins.
+ Implement getting the video/audio/visualisation elements.
+ Use input-selector for stream switching.
+ If we are asked to do visualisation, prefer to autoplug raw sinks
+ instead of sinks that accept encoded data.
+
+2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gst_play_sink_init), (gst_play_sink_dispose),
+ (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
+ (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
+ (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
+ (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
+ (gst_play_sink_set_volume), (gst_play_sink_get_volume),
+ (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
+ (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
+ (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
+ * gst/playback/gstplaysink.h:
+ Add methods to get audio/video/vis elements.
+ Add methods to set the font description for the overlay.
+ Remove properties, we're using this element with its methods only.
+ Add support for subtitles.
+ Rearrange the locking a bit to not use the object lock for protecting
+ the pipeline construction.
+ Try to use the volume and mute property on the sink when its available.
+ Implement the mute option with volume when the sink does not have a mute
+ property.
+ Only add volume element when the sink has no volume property.
+ Only do visualisations with raw audio pads.
+
+2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
+ (gst_text_overlay_init), (gst_text_overlay_set_property),
+ (gst_text_overlay_get_property), (gst_text_overlay_src_event),
+ (gst_text_overlay_text_event), (gst_text_overlay_video_event),
+ (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
+ (gst_text_overlay_change_state):
+ * ext/pango/gsttextoverlay.h:
+ Add property to configure waiting for text on the textpad or not, with
+ the default behaviour being the old one (always wait for text before
+ rendering the video). This default behaviour is usually not the best one
+ because the text stream can very sparse and could require queueing a lot
+ of video.
+ Fix the flushing and EOS handing so that we don't mix up their meaning.
+
+2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c:
+ (gst_uri_decode_bin_autoplug_factories),
+ (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
+ (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
+ (gst_uri_decode_bin_set_property),
+ (gst_uri_decode_bin_get_property), (no_more_pads_full),
+ (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
+ (proxy_autoplug_factories_signal), (make_decoder),
+ (source_new_pad), (setup_source):
+ Add a readonly source property and notify.
+ Add new lock for protecting the construction of the pipeline.
+ Keep track of the decodebins we plugged.
+ Correctly proxy the autoplug signal so that it actually continues.
+ Proxy subtitle-encoding to the decodebins.
+
+2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
+ (text_toggle_cb), (update_streams), (main):
+ Rearrange some buttons in playbin2 and make some other boxes insensitive
+ when needed.
+ Add language codes to subtitle selection boxes when we gind the right
+ tags for the streams.
+
+2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
+ (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
+ (gst_decode_bin_set_subs_encoding),
+ (gst_decode_bin_get_subs_encoding),
+ (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
+ (deactivate_free_recursive):
+ Protect caps property with the object lock.
+ Protect encoding property with the object lock.
+ Keep list of elements we added that have the subtitle-encoding property.
+ Distribute the subtitle-encoding to all of the elements when it
+ changes.
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/dvdread/stream_labels.c:
- (dvdreadsrc_get_audio_stream_labels):
- Use NULL for an empty GList instead of g_list_alloc(); fix
- memory leaks; s/LCPM/LPCM/; use g_strdup_printf() instead
- of GString (easier to bulk free later)
+ gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
+ Small debug improvement.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Fix bug in determining the sample start/stop position, we want to base
+ this decision on the fact that we are going forwards or backwards, not
+ slower or faster. This fixes some ugly resync warnings when playing at
+ very slow speeds.
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Correctly set the supported URI schemes and don't leave
+ some schemes in the middle or at the start at NULL.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix BGRA32 caps (#164209).
+2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+ tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c:
+ Make test compile without unused function/variable warnings on PPC.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_class_init):
+ * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_class_init):
+ * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
+ * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
+ * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
+ * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
+ * ext/pango/gsttextrender.c: (gst_text_render_class_init):
+ * ext/theora/theoradec.c: (gst_theora_dec_class_init):
+ * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
+ * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_class_init):
+ * gst-libs/gst/interfaces/mixertrack.c:
+ (gst_mixer_track_class_init):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init):
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_class_init):
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init):
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init):
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (preroll_unlinked):
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
+ * gst/playback/gstqueue2.c: (gst_queue_class_init):
+ * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
+ * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
+ (gst_stream_selector_class_init):
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
+ * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
+ * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init):
+ * gst/volume/gstvolume.c: (gst_volume_class_init):
+ * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
+ * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
+ * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
+ Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
+ static strings (i.e. all). This gives us less memory usage,
+ fewer allocations and thus less memory defragmentation. Depend
+ on core CVS for this. Fixes bug #523806.
+
+2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Filter http and https protocols. GIO/GVfs handles them but it's
+ impossible to implement iradio/icecast with it. Better use
+ souphttpsrc or something else for this.
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
+ If getting the file informations by a query fails try it with the
+ seek-to-end trick too.
+
+2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (gst_volume_interface_supported),
+ (gst_volume_base_init), (gst_volume_class_init),
+ (volume_process_double), (volume_process_float),
+ (volume_transform_ip), (plugin_init):
+ memset buffers to zero if we get a GAP buffer. We usually see a
+ buffer as one unit so let's handle it as one and don't care about
+ volume changes while processing one buffer.
+ Also clean up some stuff a bit.
+
+2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
+ (gst_audio_convert_create_silence_buffer),
+ (gst_audio_convert_transform):
+ Make audioconvert GAP-aware by outputting silence buffers when the
+ input has the GAP flag set. This is up to 8x faster.
+ Based on a patch by Stefan Kost. Fixes bug #517813.
+
+2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_process_double):
+ Use oil_scalarmultiply_f64_ns() for double processing when it's
+ available at compile time.
+
+2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
+ Original commit message from CVS:
+ * configure.ac:
+ Fix lrint/lrintf checks to actually work. These functions are
+ in libm on Linux at least so try to link to it.
+
+2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to development - 0.10.18.1
+ Original commit message from CVS:
+ * configure.ac:
+ Back to development - 0.10.18.1
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- alpha_mask can be RGBA/ABGR. Fixes #164265.
+=== release 0.10.18 ===
-2005-01-17 Francis Labonte <francis_labonte@hotmail.com>
+2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * po/LINGUAS:
+ * win32/common/config.h:
+ Release 0.10.18
+ Original commit message from CVS:
+ Release 0.10.18
- * ext/mpeg2dec/gstmpeg2dec.c: (crop_buffer),
- (gst_mpeg2dec_alloc_buffer):
- * ext/mpeg2dec/gstmpeg2dec.h:
- Crop if decoding size is not the actual image size (#163676).
+2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-17 Steve Baker <steve@stevebaker.org>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 0.10.17.4 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.17.4 pre-release
+
+2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
+ Original commit message from CVS:
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
+ Use GST_STR_NULL when trying to print strings that could be NULL because
+ this might crash on some platforms. See #520808.
+
+2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
+ (read_line), (gst_rtsp_connection_read_internal):
+ Generic Windows fixes that makes libgstrtsp work on Windows when
+ coupled with the new GstPoll API. See #520808.
+
+2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
+
+ ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
+ Original commit message from CVS:
+ Patch by: Milosz Derezynski <internalerror at gmail dot com>
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
+ If seeking to a new position succeeds don't simply return from
+ create() without creating a buffer. Do this only in the case
+ seeking to the new position fails. Fixes bug #523054.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (aiff_type_find),
- (svx_type_find), (sds_type_find), (ircam_type_find), (plugin_init):
- Add libsndfile typefind functions (#163309).
+ gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
+ (gst_video_format_from_rgba32_masks):
+ Fix gst_video_format_parse_caps() for RGB caps with alpha channel
+ (#522635).
+ * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
+ Add unit test for the RGB caps parsing and creation, checking for
+ internal consistency of the new API and consistency of the API with
+ the old GST_VIDEO_CAPS_* defines.
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
- * tools/gst-launch-ext-m.m:
- Add .aac, fix .m1v/.m2v (#163891).
+ gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c: Oops, revert last change
+ because -base is in freeze.
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_wait):
- Sanity check, don't wait endlessly since the clock might not
- actually run at this point (which is a deadlock). Fixes #164069.
+ gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
+ Original commit message from CVS:
+ Patch by: William M. Brack
+ * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Of course, only pause if group is done...
+ gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_selector_pad_event),
+ (gst_selector_pad_chain):
+ * gst/playback/gststreamselector.h:
+ Revert change that caused regression until a real fix is found.
+ Fixes #522203.
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Thread safety.
+ gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Rename recently added buffer types to make more sense.
+ * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
+ (gst_alsasink_write):
+ Adapt for above API changes.
+ Fixes bug #520523.
-2005-01-16 Jan Schmidt <thaytan@mad.scientist.com>
+2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
+ Original commit message from CVS:
+ * win32/common/libgstnetbuffer.def:
+ Add new symbol gst_netaddress_equal. Fixes bug #521743.
+
+2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 0.10.17.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.17.3 pre-release
+
+2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Fix duration when no clock was provided. Fixes #520300.
+
+2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
+
+ Add trivial function to compare GstNetAddress. See #520626.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester at tester ca>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ Add trivial function to compare GstNetAddress. See #520626.
+ API: GstNetBuffer::gst_netaddress_equal
+
+2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ Update mode property docs, it's deprecated now.
+
+2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/: Remove GstPollMode from gstpoll constructor.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_create):
+ * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
+ (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
+ Remove GstPollMode from gstpoll constructor.
+
+2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ 0.10.17.2 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.17.2 pre-release
+
+2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
+ Original commit message from CVS:
+ * gst/Makefile.am:
+ GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
+ them twice
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstrtp.def:
+ Add new API to the defs
+
+2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
+ Original commit message from CVS:
+ Patch by: Mersad Jelacic <mersad at axis dot com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
+ possible to specify the sample size in bits. (#509637)
+
+2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/libs/mixer.c: Add a few simple checks for the new message types.
+ Original commit message from CVS:
+ * tests/check/libs/mixer.c:
+ Add a few simple checks for the new message types.
- * ext/swfdec/gstswfdec.c: (gst_swfdec_change_state):
- Don't return state change success when the parent
- failed.
+2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
+ (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
+ (gst_mixer_message_get_type),
+ (gst_mixer_message_parse_option_changed),
+ (gst_mixer_message_parse_options_list_changed):
+ * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
+ (GST_MIXER_MESSAGE_OPTION_CHANGED),
+ (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
+ (GST_MIXER_MESSAGE_MIXER_CHANGED):
+ API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
+ and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
- * gst/avi/gstavimux.c: (gst_avimux_handle_event):
- Free events (fix memleak in #162905).
+2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-15 Gergely Nagy <algernon@bonehunter.rulez.org>
+ gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
+ (gst_mixer_options_get_values):
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
+ (_GstMixerOptions), (_GstMixerOptionsClass):
+ API: add GstMixerOptions::get_values vfunc (#519906)
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
+ Original commit message from CVS:
+ * configure.ac:
+ Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
+ plug-ins are included/excluded. (#498222)
+
+2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinder for IMelody files, using audio/x-imelody.
+ See bug #519516.
+
+2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
+ * ext/alsa/gstalsasink.c: (set_hwparams):
+ * ext/alsa/gstalsasrc.c: (set_hwparams):
+ * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
+ * ext/ogg/gstoggmux.h:
+ * ext/ogg/gstogmparse.c:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ (gst_missing_uri_sink_message_new),
+ (gst_missing_element_message_new),
+ (gst_missing_decoder_message_new),
+ (gst_missing_encoder_message_new):
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ (gst_rtcp_packet_bye_get_reason):
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/playback/test.c: (gen_video_element), (gen_audio_element):
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * sys/v4l/gstv4lelement.c:
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
+ * sys/v4l/v4l_calls.c:
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
+ (gst_v4lsrc_try_capture):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
+ (gst_ximagesink_ximage_new):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
+ (gst_xvimagesink_xvimage_new):
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/audioresample.c:
+ (fail_unless_perfect_stream):
+ * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
+ * tests/check/elements/decodebin.c:
+ * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
+ (setup_gdpdepay_streamheader):
+ * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
+ (setup_gdppay_streamheader):
+ * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
+ * tests/check/elements/multifdsink.c: (setup_multifdsink):
+ * tests/check/elements/textoverlay.c:
+ * tests/check/elements/videorate.c: (setup_videorate):
+ * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
+ * tests/check/elements/volume.c: (setup_volume):
+ * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
+ * tests/check/elements/vorbistag.c:
+ * tests/check/generic/clock-selection.c:
+ * tests/check/generic/states.c: (setup), (teardown):
+ * tests/check/libs/cddabasesrc.c:
+ * tests/check/libs/video.c:
+ * tests/check/pipelines/gio.c:
+ * tests/check/pipelines/oggmux.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ (simple_launch_lines_suite):
+ * tests/check/pipelines/streamheader.c:
+ * tests/check/pipelines/theoraenc.c:
+ * tests/check/pipelines/vorbisdec.c:
+ * tests/check/pipelines/vorbisenc.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c: (query_positions_elems),
+ (query_positions_pads):
+ * tests/icles/stress-xoverlay.c: (myclock):
+ Correct all relevant warnings found by the sparse semantic code
+ analyzer. This include marking several symbols static, using
+ NULL instead of 0 for pointers and using "foo (void)" instead
+ of "foo ()" for declarations.
+ * win32/common/libgstrtp.def:
+ Add gst_rtp_buffer_set_extension_data to the symbol definition file.
+
+2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
+
+ gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
+ Original commit message from CVS:
+ Patch by: José Alburquerque <jaalburqu svn gnome org>
+ * gst/playback/gstplaybin2.c:
+ Make the function signature of the _get_*_tags() functions match
+ the signature of the vfuncs they implement, ie. return a
+ GstTagList rather than a GstStructure, which is more correct,
+ even if one is typedef'ed to the other (#518940).
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- Fix for depth = 15. Fixes #161675.
+2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Don't include unix headers unconditionally (fixes #518037).
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Set FPS correctly, even for webcams and the like.
- * sys/v4l/v4l_calls.c: (gst_v4l_set_chan_norm):
- Don error on setting while capturing.
+2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
+ Original commit message from CVS:
+ * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
+ (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
+ (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
+ (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
+ (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
+ (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
+ (gst_video_format_is_packed), (video_format_is_packed):
+ Add unit test that makes sure that the strides, offsets and
+ sizes returned for the various YUV formats by the new video API
+ match the old reference implementation in videotestsrc.
- * ext/dv/gstdvdec.c:
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
- I'm a bad boy. using /1001. to force C to do float division
- and not integer division (as it did in my last commit)
- Thanks to David I. Lehn for pointing this mistake.
+2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
+ (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
+ (gst_video_format_is_rgb), (gst_video_format_is_yuv),
+ (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
+ (gst_video_format_get_pixel_stride),
+ (gst_video_format_get_component_width),
+ (gst_video_format_get_component_height),
+ (gst_video_format_get_component_offset), (gst_video_format_get_size):
+ * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
+ (GST_VIDEO_FORMAT_Y42B):
+ API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Revert Johan's 1.35->1.36 since it breaks compat.
+2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
+ YV12 is I420 with swapped components 1 and 2, so the offset of
+ component 1 for I420 should be the offset for component 2 for YV12
+ and vice versa.
- * ext/dv/gstdvdec.c:
- * ext/libfame/gstlibfame.c:
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
- replace framerate aproximations by their real value
- (24000/1001, 30000/1001, 60000/1001)
- Finish fixing bug #164049
+2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
+
+ sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
+ Original commit message from CVS:
+ * sys/v4l/gstv4lelement.c:
+ Add missing semicolon to fix indentation.
+
+2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
+ Original commit message from CVS:
+ 2008-02-29 Julien Moutte <julien@fluendo.com>
+ * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
+ (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
+ detect
+ if we can do SPDIF output.
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
+ (gst_alsasink_prepare), (gst_alsasink_close),
+ (gst_alsasink_write):
+ * ext/alsa/gstalsasink.h: Initial support for SPDIF.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_parse_caps):
+ * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
+ types
+ to support AC3, EC3 and IEC958 buffers.
-2005-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggmux.c:
- eos/bos debugging
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- improve reusability of elements after state changes and errors
- make multifdsink throw away streamheaders when receiving new ones
+ gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
+ (gst_mixer_message_parse_mute_toggled),
+ (gst_mixer_message_parse_record_toggled),
+ (gst_mixer_message_parse_volume_changed),
+ (gst_mixer_message_parse_option_changed):
+ De-cruft and fix message type assertions (NULL is not a really
+ valid mixer message type string).
-2005-01-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_vis_src_negotiate):
+ When negotiating, actually start from a format that we can support
+ instead of from the too generic template.
+
+2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Enable vis setting.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
+ Enable vis setting.
+ * gst/playback/gstplaysink.c: (gst_play_sink_init),
+ (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
+ (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
+ (gen_vis_chain):
+ Implement vis switching while playing.
+
+2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/riff/riff-media.c: Add Dirac mapping
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
+
+2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/tcp/Makefile.am:
+ * gst/tcp/fdsetstress.c:
+ * gst/tcp/gstfdset.c:
+ * gst/tcp/gstfdset.h:
+ Removed fdset and stress test, they are now known as GstPoll in
+ core.
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
+ (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
+ (gst_multi_fd_sink_handle_client_write),
+ (gst_multi_fd_sink_queue_buffer),
+ (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
+ (gst_multi_fd_sink_stop):
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
+ (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
+ (gst_tcp_gdp_read_caps):
+ * gst/tcp/gsttcp.h:
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
+ (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
+ (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
+ * gst/tcp/gsttcpclientsink.h:
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
+ (gst_tcp_client_src_create), (gst_tcp_client_src_start),
+ (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
+ * gst/tcp/gsttcpclientsrc.h:
+ * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
+ (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
+ * gst/tcp/gsttcpserversink.h:
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
+ (gst_tcp_server_src_create), (gst_tcp_server_src_start),
+ (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
+ * gst/tcp/gsttcpserversrc.h:
+ Port to GstPoll. See #505417.
+
+2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
- Fix for if items are already in list...
+ * ChangeLog:
+ Patch Changelog a bit to give credit and refer to the relevant bug.
+ Original commit message from CVS:
+ Patch Changelog a bit to give credit and refer to the
+ relevant bug.
+
+2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
+ (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
+ (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
+ (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
+ (gst_rtsp_connection_flush):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Use GstPoll for the rtsp connection.
+
+2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
+ (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
+ Add combo box for visualisations, populate it with a factory list
+ of all visualisation plugins, configure vis plugin instance in
+ playbin2.
+
+2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
+ Original commit message from CVS:
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Add check for RTP buffer defaults, padding and marker bit API.
+
+2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/sha1.c: (sha_transform):
+ Use memcpy() instead of upcasting a byte array to long *. This
+ fixes an unaligned memory access, resulting in SIGBUS on IA64.
+ This should be ported to GCheckSum once we can use GLib 2.16.
+ Partially fixes bug #500833.
+
+2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
+ Push tag event after the newsegment event. Log the pointer of
+ the buffer we're actually going to push rather than the buffer
+ we're feeding to _make_metadata_writable().
-2005-01-12 Benjamin Otte <otte@gnome.org>
+2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/adder/gstadder.c: (gst_adder_loop):
- fix adder a bit so it doesn't screw up with events as much anymore
+ gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Comment smoke typefinder for now. The smokedec plugin needs one
+ frame per buffer but we have no parser yet, thus it simply crashes
+ in most situations.
-2005-01-12 Jan Schmidt <thaytan@mad.scientist.com>
+2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_link),
- (pixbufscale_scale), (gst_pixbufscale_chain):
- * ext/gdk_pixbuf/pixbufscale.h:
- Incorporate changes from Tim-Philipp Mueller <t.i.m@orange.net>
- to ensure rowstrides are calculated the same way as
- ffmpegcolorspace
- Use gst_buffer_stamp instead of copying TIMESTAMP manually, so
- that we pick up duration and offset also.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinder for the smoke video codec. Copied from the jpeg plugin.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/avi/gstavimux.c: (gst_avimux_class_init),
- (gst_avimux_pad_unlink), (gst_avimux_release_pad):
- Reusability fixes.
+ gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
+ (plugin_init):
+ Add midi typefinder, copied from the timidity plugin.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_update),
- (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
- (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
- (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
- Update flags when requested.
+ Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
+ Original commit message from CVS:
+ Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
+ * gst/subparse/gstsubparse.c: (parse_mdvdsub):
+ * tests/check/elements/subparse.c: (test_microdvd_with_italics),
+ (subparse_suite):
+ Forward slashes at the beginning and end of a line also signify
+ italics (Fixes: #518162).
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
- Fix dmix.
+ tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
+ Original commit message from CVS:
+ * tests/check/gst-plugins-base.supp:
+ Add a suppression for a cached value in GIO that wasn't moved
+ while moving gio from -bad to -base.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (gst_play_base_bin_dispose),
- (probe_triggered), (new_decoded_pad), (gen_source_element),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (group_switch), (remove_sinks), (setup_sinks),
- (gst_play_bin_change_state):
- Implement group-switch signal for use in apps to clear metadata
- cache, clean up subtitle, add suburi property instead of # hack,
- some error-out fixes.
+ configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
+ Original commit message from CVS:
+ Patch by: Brian Cameron <brian dot cameron at sun dot com>
+ * configure.ac:
+ Don't hardcode -Wall and -Werror for configure checks, this fails
+ with non-GCC compilers. Fixes bug #517991.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Debug.
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_grab_frame):
- If we got a state change in the _get handler, don't return success.
+ gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_my_output_message),
- (gst_jpegdec_my_emit_message), (gst_jpegdec_init):
- Make jpegdec quiet on MJPEG decoding
- * gst/asfdemux/README:
- Fix mimetypes for MJPEG and H263
+ ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ (gst_gnome_vfs_sink_handle_event):
+ Return FALSE when seeking for a new segment fails instead
+ of silently ignoring the failure and appending every buffer
+ that comes for the new segment.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/theora/theoradec.c: (theora_dec_chain):
- Fix broken code generation by gcc by swapping arguments.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix \n in debug.
+ gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (find_property),
+ (gst_play_sink_find_property), (gen_video_chain),
+ (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
+ Recursively search the sink element for a last-frame property so that we
+ can also find the property in autovideosink and friends that don't
+ always proxy the internal sink properties.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * TODO:
- delete this file, it is by far outdated
- * ext/alsa/gstalsa.1: remove
- * ext/alsa/gstalsa.c: (add_rates), (add_channels), (gst_alsa_caps),
- (gst_alsa_check_sample_rates), (gst_alsa_rates_probe),
- (gst_alsa_get_caps):
- Add HW probing for supported sample rates. Fixes #161704
+ gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
+ (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
+ (gst_audio_set_structure_channel_positions_list),
+ (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
+ (gst_audio_fixate_channel_positions):
+ Fix confusing terminology in docs and code: structure fields are
+ 'fields' and not 'properties'.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Don't crash, biatch! :).
+ gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions), (add_list_to_struct):
+ Give more useful warning messages if one of the channel
+ layout enums passed to us is invalid and if the "channels"
+ field in the caps has a GType we don't expect.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/musepack/gstmusepackreader.cpp:
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
- Some work on tags - still doesn't work in playbin...
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Handle events...
+ gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ Fix typo in docs blurb.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
- Also shove tags on kid pads.
+ gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
+ Original commit message from CVS:
+ 2008-02-19 Julien Moutte <julien@fluendo.com>
+ Patch by: Josep Torra Valles <josep@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
+ typefind lookup to fix typefinding on HD clips.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
- Don't bail on unknown events.
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Don't crash on events before negotiation.
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- Send tags on pads, too.
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_request_new_pad):
- Forward events on first pad if no input was selected yet.
+ gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
+ Original commit message from CVS:
+ * gst/playback/gstscreenshot.c:
+ * gst/playback/gstscreenshot.h:
+ Fix up copyright (I rewrote the GStreamer-0.10 code for
+ this from scratch back in the days).
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Add screenshot conversion code from totem.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
+ (create_element), (gst_play_frame_conv_convert):
+ * gst/playback/gstscreenshot.h:
+ Add screenshot conversion code from totem.
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
+ (gst_play_bin_class_init), (gst_play_bin_convert_frame),
+ (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
+ Implement frame property to get a color-unconverted snapshot.
+ Implement convert-frame action signal to get a converted snapshot image.
+ Configure connection speed in uridecodebin.
+ Document some more properties.
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_get_last_frame):
+ * gst/playback/gstplaysink.h:
+ Use last-buffer property of the video sink to get a video snapshot.
+ * tests/examples/seek/seek.c: (shot_cb), (main):
+ Add snapshot button for playbin2 and use the frame property to save the
+ frame as a png in the current directory.
+
+2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
+ Original commit message from CVS:
+ Patch by: Josep Torra Valles <josep at fluendo dot com>
+ * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
+ (plugin_init):
+ Add typefinding support for h264 elementary streams.
+ Fixes bug #517420.
- * gst/playback/gstplaybasebin.c: (setup_substreams):
- Don't disable streamtype if the stream doesn't exist, since
- then playing a video after audio will disable both and nothing
- will happen. Fixes the testsuite.
+2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Require CVS of core for new API in collectpads.
+ Original commit message from CVS:
+ * configure.ac:
+ Require CVS of core for new API in collectpads.
+ * gst/adder/gstadder.c:
+ Use new API to make adder sparse stream aware.
- * sys/v4l/gstv4lxoverlay.c: (gst_v4l_xoverlay_interface_init),
- (gst_v4l_xoverlay_set_xwindow_id):
- * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_interface_init),
- (gst_v4l2_xoverlay_set_xwindow_id):
- Add debug categories, fix overlay disabling.
+2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
+ (no_more_pads_cb):
+ Get the object data correct so that we can remove our channels
+ correctly.
+ * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
+ (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_request_pad):
+ Add option to disable async behaviour in the sinks when possible. This
+ makes it possible to avoid an audio queue when dealing with
+ visualisations.
+ Add option to add a queue for the audio path.
+ * tests/examples/seek/seek.c: (clear_streams), (update_streams),
+ (main):
+ Disable the vis checkbox to match the defaults of playbin2.
+ Only get the stream info when we need to.
+
+2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Don't use async operations as they require a running main loop.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
+ (gst_gio_base_sink_set_stream):
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
+ (gst_gio_base_src_set_stream):
+ * ext/gio/gstgiosink.c: (gst_gio_sink_start):
+ * ext/gio/gstgiosrc.c: (gst_gio_src_start):
+ Don't use async operations as they require a running main loop.
+ This makes us block again when closing streams and unable
+ to mount the enclosing volume of an URI if it isn't yet.
+
+2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
+ (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
+ (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_request_pad):
+ Move tee in front of the audio and vis pipelines.
+ Add queue for audio for now.
+ Add visualisation support.
+ * tests/examples/seek/seek.c: (main):
+ Visualisation is by default disabled.
+
+2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Improve debugging a bit.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (close_stream_cb):
+ * ext/gio/gstgiobasesrc.c: (close_stream_cb):
+ Improve debugging a bit.
+ * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
+ * ext/gio/gstgiosrc.h:
+ Try to mount the enclosing volume of a GFile if it isn't mounted
+ yet. This requires us to wait for an async operation to finish, done
+ with an nested GMainLoop. Authentication is not supported yet, will
+ come later.
+
+2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Add mute property.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_property), (gst_play_bin_get_property),
+ (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
+ (gst_play_sink_get_mute), (gen_audio_chain):
+ * gst/playback/gstplaysink.h:
+ Add mute property.
+ * gst/playback/gststreamselector.c: (gst_selector_pad_event),
+ (gst_selector_pad_chain):
+ * gst/playback/gststreamselector.h:
+ Make sure we forward the event only once.
+ * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
+ Add and implement the mute button for playbin2.
+
+2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
+
+ ext/alsa/gstalsasink.c: Add some more debug info.
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
+ Add some more debug info.
+ Make sure we never return a negative delay. Fixes #516246.
+
+2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
+ Revert patch that makes the sink hold the object lock when
+ calling snd_pcm_delay(), since it breaks playback for me.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Add some seek flags when changing rate.
+ Original commit message from CVS:
+ 2008-02-12 Julien Moutte <julien@fluendo.com>
+ * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
+ some seek flags when changing rate.
+
+2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer):
+ Fix potential leaks.
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
+ Fix leak when there is no function configured.
+
+2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
+ Original commit message from CVS:
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
+ (gst_v4lsrc_buffer_finalize):
+ Correctly chain up the finalize method.
+
+2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
+ Original commit message from CVS:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ Add documentation and example code for giostreamsink/giostreamsrc.
+ * tests/check/pipelines/gio.c: (GST_START_TEST):
+ Ask the GMemoryOutputStream for the data instead of assuming that
+ the pointer to the data stayed the same. It could've been realloc'ed.
+
+2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
+ Original commit message from CVS:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ Make the documentation of giosink/giosrc complete, large parts
+ are based on the gnomevfssink/gnomevfssrc docs.
+
+2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/: Add the GIO documentation again and while at that run make update.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ Add the GIO documentation again and while at that run make update.
+
+2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be...
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
+ * ext/alsa/gstalsasink.c: (set_swparams):
+ * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
+ Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
+ against libasound >= 1.0.16, since it's been deprecated in
+ 0.10.16, and alignment is always 1 then, apparently. (#512899)
- * ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_get_caps):
- * ext/alsa/gstalsa.h:
- Add HW probing for period_count/size and buffer_size MIX/MAX
- Adjust default/user defined value if out of bounds
- Should fix bug #162024
+2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_audio_element):
+ * gst/playback/gstplaysink.c: (gen_audio_chain):
+ Handle case where we can't create the volume element a bit
+ better (#514307).
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event):
- Fix warning (#161191).
+2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ ext/gnomevfs/: Add support for https protocol. Fixes #510229.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
+ * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
+ Add support for https protocol. Fixes #510229.
- * ext/dvdread/stream_labels.c:
- (dvdreadsrc_get_audio_stream_labels):
- Fix warning (init the good variable in switch default)
+2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
-2005-01-09 Koop Mast <kwm@rainbow-runner.nl>
+ ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
+ Original commit message from CVS:
+ 2008-02-11 Julien Moutte <julien@fluendo.com>
+ Patch by: Alan Peevers <peeves@pacbell.net>
+ * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
+ lock when calling alsa methods.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tta/gstttaparse.c: (gst_tta_src_event):
- Fix gcc-2.95 compile (#163485).
+ gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ Bump rank of jpeg and png typefinders, which will return maximum
+ probability in the most common cases (thus short-circuiting more
+ expensive typefinders like the mp3 one for these two quite common
+ image types).
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
+ Original commit message from CVS:
+ * ext/theora/theoraparse.c:
+ Fix long description of the theora parser to be more verbose than just
+ the type name.
+
+2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
+
+ sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
+ Original commit message from CVS:
+ Patch by: Branko Čibej <brane at xbc dot nu>
+ * sys/xvimage/xvimagesink.c:
+ Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
+ Fixes bug #515654.
+
+2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
+
+ gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ Set is_dynamic as True if there are elements with both request
+ and sometimes src pad templates instead of breaking out when it
+ finds the first pad template that is a src.
+
+2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
+ (update_streams), (video_combo_cb), (audio_combo_cb),
+ (text_combo_cb), (volume_spinbutton_changed_cb), (main):
+ Add some stream switching and volume gui for playbin2.
+
+2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
+ Original commit message from CVS:
+ * gst/playback/gstplay-marshal.list:
+ Added marshal for streamselector Tags.
+ * gst/playback/gstplaybasebin.c: (set_active_source):
+ Streamselector now selects pads based on the pad object instead of its
+ name.
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (init_group), (gst_play_bin_init), (get_group), (get_tags),
+ (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
+ (gst_play_bin_get_text_tags),
+ (gst_play_bin_set_current_video_stream),
+ (gst_play_bin_set_current_audio_stream),
+ (gst_play_bin_set_current_text_stream),
+ (gst_play_bin_set_property), (gst_play_bin_get_property),
+ (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
+ Remove option to mute streams with the current-a/v/t property, we have
+ this functionality in the flags.
+ Add signals to notify when the number of A/V/T channels changed.
+ Add action signals to get tags for the A/V/T streams.
+ Implement setting the current A/V/T stream.
+ Rearrange some things to simplify stream selection.
+ Implement volume.
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
+ (gst_play_sink_get_volume), (gst_play_sink_set_property),
+ (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
+ (activate_vis), (gst_play_sink_reconfigure):
+ * gst/playback/gstplaysink.h:
+ Add and implement volume setting methods.
+ * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
+ (gst_selector_pad_finalize), (gst_selector_pad_get_property),
+ (gst_selector_pad_event), (gst_stream_selector_class_init),
+ (gst_stream_selector_init), (gst_stream_selector_finalize),
+ (gst_stream_selector_set_property),
+ (gst_stream_selector_get_property),
+ (gst_stream_selector_get_linked_pad),
+ (gst_stream_selector_request_new_pad):
+ * gst/playback/gststreamselector.h:
+ Add pad properties for tags and status of pads.
+ Keep tags on pads.
+ Make active pad selection based on pad object instead of name.
+
+2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
+ Original commit message from CVS:
+ * configure.ac:
+ Revert last change as we now check in gtk-doc.m4 for sed.
- * configure.ac:
- * ext/flac/gstflacenc.c: (gst_flacenc_init),
- (gst_flacenc_seek_callback), (gst_flacenc_write_callback),
- (gst_flacenc_tell_callback), (gst_flacenc_chain),
- (gst_flacenc_change_state):
- * ext/flac/gstflacenc.h:
- Update for API change in flac-1.1.1. Update requirement in
- configure.ac. Fixes #162974.
+2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Find and subst SED when building the docs.
+ Original commit message from CVS:
+ * configure.ac:
+ Find and subst SED when building the docs.
+
+2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
+ Original commit message from CVS:
+ 2008-02-08 Julien Moutte <julien@fluendo.com>
+ * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
+ (main): Make sure bus signals are reconnected when pressing STOP
+ and then PLAY again for a parse launch pipeline. Fix a ref leak
+ on the bus.
+ * win32/common/config.h: Updated.
+
+2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
+ Original commit message from CVS:
+ * configure.ac:
+ Make DISABLE_DEPRECATED defined *only* during CVS, not during
+ pre-releases or releases.
+
+2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gio/Makefile.am:
+ Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
+ reporting
+
+2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ Add the headers which need scanning for the GIO plugin. The rest of
+ the docs still need migrating.
+
+2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Add gio in a few more places.
+ Original commit message from CVS:
+ * ext/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/.cvsignore:
+ Add gio in a few more places.
+
+2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Move gio plugin from -bad and mark as experimental.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/Makefile.am:
+ * tests/check/Makefile.am:
+ Move gio plugin from -bad and mark as experimental.
+
+2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ Comment out a couple of other things which break the build when
+ GST_DISABLE_DEPRECATED isn't on but -Werror is.
+
+2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Fix pbutils header.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/playback/gstplaybasebin.c: (group_destroy):
- Remove hack to get rid of assert and get rid of unlinked
- signals properly.
+ * gst-plugins-base.spec.in:
+ commit spec file update which includes all the split .pc files
+ Original commit message from CVS:
+ commit spec file update which includes all the split .pc files
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Set source to NULL so that resources are free'ed. Fixes issues
- with playback of CDDA and similar device-accessing things.
+ gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
+ Fix compiler warning.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
- * testsuite/embed/Makefile.am:
- test->noinst, fix make test in buildbot.
+ gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
+ Clear the addrinfo struct using memset. Fixes #514937.
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/dvdread/stream_labels.c: new file
- * ext/dvdread/stream_labels.h: new file
- * ext/dvdread/Makefile.am:
- * ext/dvdread/dvdreadsrc.c: (_seek_title):
- Extract audio stream label from DVD IFO files.
- It only dump them on the console for now, still have to
- make playbin aware of them.
+ gst/tcp/gstfdset.h: Remove unused field to same some memory.
+ Original commit message from CVS:
+ * gst/tcp/gstfdset.h:
+ Remove unused field to same some memory.
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ Mark action signals as such.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix hanging subs.
+ ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (_theora_granule_frame),
+ (_inc_granulepos):
+ Increment granulepos for new-bitstream versions appropriately.
+ Fixes #514623.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (gen_preroll_element), (remove_groups), (setup_subtitle),
- (gen_source_element), (setup_source):
- * gst/playback/gstplaybasebin.h:
- Multiple .sub files is just a stupid idea... Fix some threading
- mistakes. Interestingly, external .sub files cause playbin to
- hang, I don't know why... Parsing fixes contributed by Fran??ois
- Kooman <fkooman@tuxed.net>.
+ tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (do_seek),
+ (rate_spinbutton_changed_cb), (update_streams), (main):
+ Remove obsolete stream_time reset after flushing seek, core does that
+ automatically now.
+ Improve accuracy of speed spinbutton.
+ Only do playbin2 stuff when we actually use it.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * testsuite/embed/Makefile.am:
- Fix buildbot.
+ tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Revert previous change of the test environment's GST_PLUGIN_PATH.
+ The problem is not with the plugins, but with element factories
+ and only occurs if elements are split out from existing plugins
+ or if plugins change name (see #512740).
-2005-01-09 Gergely Nagy <algernon@bonehunter.rulez.org>
+2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix the tests environment's GST_PLUGIN_PATH: we want the directory
+ with the core's plugins first and our local build directories last,
+ since we might be building against an installed core, and that
+ core's plugin directory may contain older or other versions of
+ our own -base plugins, but we really do want to test our local
+ ones (if there are multiple plugins or element factories with the
+ same name, those inspected last will trump those read in earlier).
+ Fixes #512740 for the most part.
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
- (gst_pngenc_init), (gst_pngenc_chain), (gst_pngenc_get_property),
- (gst_pngenc_set_property):
- * ext/libpng/gstpngenc.h:
- Add compression level property (#163323).
+2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Use gmtime_r if available as gmtime is not MT-safe.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Use gmtime_r if available as gmtime is not MT-safe.
+ Fixes bug #511810.
+
+2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Cast glong to time_t as time_t might have a different type on
+ other platforms, like FreeBSD, and we get a compiler warning
+ otherwise. Fixes bug #511825.
+
+2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (get_group), (get_n_pads), (gst_play_bin_get_property),
+ (pad_added_cb), (no_more_pads_cb), (perform_eos),
+ (autoplug_select_cb), (deactivate_group):
+ Remove stream-info, we going for something easier.
+ Refactor getting the current group.
+ Implement getting the number of audio/video/text streams.
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_class_init), (gst_stream_selector_init),
+ (gst_stream_selector_get_property),
+ (gst_stream_selector_request_new_pad),
+ (gst_stream_selector_release_pad):
+ * gst/playback/gststreamselector.h:
+ Add property for number of pads.
+ * tests/examples/seek/seek.c: (set_scale), (update_flag),
+ (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
+ (text_toggle_cb), (update_streams), (msg_async_done),
+ (msg_state_changed), (main):
+ Block slider callback when updating the slider position.
+ Add gui elements for controlling playbin2.
+ Add callback for async_done that updates position/duration.
+
+2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/plugins/: First round of plugin docs cleansups.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ First round of plugin docs cleansups.
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Regenerate.
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ Add header for oggmux. the c-file needs a doc blob still.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ Add gst_rtp_buffer_set_extension_data()
+ Original commit message from CVS:
+ Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_set_extension_data):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
+ Add gst_rtp_buffer_set_extension_data()
+ Add a unit test for this addition. Fixes #511478.
+ API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
+
+2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
+ Really clean up the queue instead of just unreffing all buffers
+ in it.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
+ (gst_app_src_class_init), (gst_app_src_init),
+ (gst_app_src_dispose), (gst_app_src_finalize):
+ Fix dispose/finalize.
+
+2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (close_stream_cb),
+ (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
+ (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
+ * ext/gio/gstgiobasesrc.c: (close_stream_cb),
+ (gst_gio_base_src_stop), (gst_gio_base_src_create),
+ (gst_gio_base_src_set_stream):
+ Use async variants of the close stream functions to prevent blocking
+ for a long time there and add some more sanity checks for a correct
+ stream.
- * configure.ac:
- * examples/capsfilter/capsfilter1.c: (main):
- * examples/seeking/spider_seek.c: (make_spider_pipeline):
- * ext/dvdread/Makefile.am:
- * ext/dvdread/demo-play:
- * ext/dvdread/demo-play.c:
- * gconf/gstreamer.schemas.in:
- * gst-libs/gst/gconf/gconf.c:
- * sys/v4l/TODO:
- * testsuite/Makefile.am:
- * testsuite/embed/Makefile.am:
- * testsuite/embed/embed.c: (cb_expose), (main):
- Remove all references to xvideosink, fix examples (#140845).
- * gst/playback/gstplaybasebin.c: (group_destroy):
- Apparently, disposal does not unlink - so do explicitely.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Add debug.
-
-2005-01-09 Maciej Katafiasz <mathrick@freedesktop.org>
-
- * README: fix PKG_CONFIG_PATH instructions, what was there
- previously was breaking default search path, not nice.
- Fixes #163358
-
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_init),
- (gst_audioscale_chain):
- %#^@^#@^@#^#@^#@^@#^@#^@#^@#^#@^#@^#@^@#^#@ fix seeking
- when resampling - how the ^@$^!@^! is this possible?!?
-
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state):
- Reset variables on READY.
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad),
- (gst_matroska_mux_loop):
- Require data before writing header.
-
-2005-01-09 Francis Labonte <francis_labonte@hotmail.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstmad.c: (gst_mad_chain):
- Don't call mad_stream_sync() directly after recovering sync.
- Fixes #151661.
-
-2005-01-09 Martin Eikermann <meiker@upb.de>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init),
- (snapshot_handler), (gst_snapshot_sinkconnect),
- (gst_snapshot_chain):
- Allocate resources when required, fix recursive signal emission
- and fix caps. Fixes #161667.
-
-2005-01-09 Gergely Nagy <algernon@bonehunter.rulez.org>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/libpng/gstpngdec.c: (gst_pngdec_src_getcaps),
- (gst_pngdec_chain):
- Handle only 8-bppc (bits-per-pixel-component) images, better
- error handling and correct strides. Fixes #163177.
- * ext/libpng/gstpngenc.c: (gst_pngenc_sinklink),
- (gst_pngenc_chain):
- Better error handling. Fixes #163348.
-
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_get_type),
- (dvdnavsrc_uri_get_type), (dvdnavsrc_uri_get_protocols),
- (dvdnavsrc_uri_get_uri), (dvdnavsrc_uri_set_uri),
- (dvdnavsrc_uri_handler_init):
- Add DVD-nav URI (dvdnav://) for Totem testing purposes.
- * gst/playback/gstplaybasebin.c: (gen_source_element):
- Add MMS to streaming URIs.
-
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_navigation_send_event):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Check for pad availability before sending event.
+2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * gst-plugins.spec.in:
- Add subparse.
+=== release 0.10.17 ===
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- Since we use functions from CVS, up requirement.
-
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ Release 0.10.17
+ Original commit message from CVS:
+ Release 0.10.17
+
+2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ Also remove the conditional registration of the signals
+ that disappeared with the ABI change in 0.10.14
+
+2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Revert patch to gstrtspconnection.c for brown paper bag
+ release of -base. Re-opens: #511825
+
+2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Change the way these deprecated function pointers are removed
+ so that the compiled ABI is unconditionally smaller. This
+ sets in stone an ABI break that actually occurred when the
+ things were deprecated in 0.10.14, which seems to be the best
+ fix as the only known users are oss-mixer and sunaudio-mixer in
+ gst-plugins-good.
+ Fixes: #513018
+
+2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Change the way these deprecated function pointers are removed
+ so that the compiled ABI is unconditionally smaller. This
+ sets in stone an ABI break that actually occurred when the
+ things were deprecated in 0.10.14, which seems to be the best
+ fix as the only known users are oss-mixer and sunaudio-mixer in
+ gst-plugins-good.
+
+2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
+ Original commit message from CVS:
+ * win32/common/libgstpbutils.def:
+ Export the two new _get_type() functions which are needed
+ by the python bindings.
- * gst/playback/Makefile.am:
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (group_destroy), (group_commit), (group_is_muted),
- (gen_preroll_element), (add_stream), (unknown_type),
- (probe_triggered), (preroll_unlinked), (mute_stream),
- (silence_stream), (new_decoded_pad), (setup_substreams),
- (setup_source), (get_active_source), (mute_group_type),
- (muted_group_change_state), (set_active_source),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (play_base_eos), (gst_play_base_bin_change_state):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (add_sink), (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
- (gst_stream_info_dispose), (stream_info_mute_pad),
- (stream_info_change_state), (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c: (gst_stream_selector_get_type),
- (gst_stream_selector_base_init), (gst_stream_selector_class_init),
- (gst_stream_selector_init), (gst_stream_selector_dispose),
- (gst_stream_selector_get_linked_pad),
- (gst_stream_selector_get_caps), (gst_stream_selector_link),
- (gst_stream_selector_get_linked_pads),
- (gst_stream_selector_request_new_pad), (gst_stream_selector_chain):
- * gst/playback/gststreamselector.h:
- Adding stream selection support plus required properties for
- applications to use this. Fully fixes #100931.
-
-2005-01-08 Benjamin Otte <otte@gnome.org>
-
- * gst/games/gstpuzzle.c: (nav_event_handler):
- - handle nav events differently: forward every event no matter if it
- was handled or not.
- - translate events
- You can now cheat by using navigationtest ! puzzle and moving the
- mouse close to the edge of a tile. ;)
-
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * configure.ac:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_new):
- * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_get_type),
- (gst_ogm_text_parse_base_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_get_sink_querytypes), (gst_ogm_parse_sink_convert),
- (gst_ogm_parse_sink_query), (gst_ogm_parse_chain),
- (gst_ogm_parse_plugin_init):
- * ext/pango/gsttextoverlay.c: (gst_textoverlay_linkedpads),
- (gst_textoverlay_link), (gst_textoverlay_getcaps),
- (gst_textoverlay_event), (gst_textoverlay_video_chain),
- (gst_textoverlay_loop), (gst_textoverlay_init), (plugin_init):
- * ext/pango/gsttextoverlay.h:
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
- (gst_matroska_demux_handle_seek_event),
- (gst_matroska_demux_sync_streams),
- (gst_matroska_demux_parse_blockgroup),
- (gst_matroska_demux_subtitle_caps),
- (gst_matroska_demux_plugin_init):
- * gst/matroska/matroska-ids.h:
- * gst/playback/gstdecodebin.c: (close_pad_link):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (gen_preroll_element), (remove_groups), (add_stream),
- (new_decoded_pad), (setup_subtitles), (gen_source_element),
- (setup_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gen_text_element), (setup_sinks):
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c: (gst_subparse_get_type),
- (gst_subparse_base_init), (gst_subparse_class_init),
- (gst_subparse_init), (gst_subparse_formats),
- (gst_subparse_eventmask), (gst_subparse_event),
- (gst_subparse_handle_event), (convert_encoding), (get_next_line),
- (parse_mdvdsub), (parse_mdvdsub_init), (parse_subrip),
- (parse_subrip_deinit), (parse_subrip_init), (parse_mpsub),
- (parse_mpsub_deinit), (parse_mpsub_init),
- (gst_subparse_buffer_format_autodetect),
- (gst_subparse_format_autodetect), (gst_subparse_loop),
- (gst_subparse_change_state), (gst_subparse_type_find),
- (plugin_init):
- * gst/subparse/gstsubparse.h:
- * gst/typefind/gsttypefindfunctions.c: (ogmtext_type_find),
- (plugin_init):
- Add subtitle support, .sub parser (supports SRT and MPsub),
- OGM text support, Matroska UTF-8 text support, deadlock fixes
- all over the place, subtitle awareness in decodebin/playbin
- and some fixes to textoverlay to handle subtitles in a stream
- correctly. Fixes #100931.
-
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Check for pad availability before doing a query on it.
-
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
-
- * ext/dv/gstdvdec.c:
- really fix bpp24/32 dvdec caps (classic rgba indeed)
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- don't send text tags if they are empty (bis repetita)
-
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
-
- * ext/dv/gstdvdec.c:
- remove unneeded comment from dvdec
- (related to DV 4CC codes in AVI files)
- moved them in gstreamer/docs/random/mimetypes
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- don't send text tags if they are empty
- fix mem leak on error path
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (img_get_alpha_info):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- adds BGR32 and BGRA32 to ffmpegcolorspace
- (still bad colors, fixing it on next commit)
- helps with dvdec outputing BGR32
+ gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Cast glong to time_t as time_t might have a different type on
+ other platforms, like FreeBSD, and we get a compiler warning
+ otherwise. Fixes bug #511825.
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/dv/gstdvdec.c:
- Fix audio caps i just broke (missing ',')
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_get_type),
- (gst_matroska_mux_reset):
- Fix typo + add FIXME about old "x-gst-metadata" crap
+ gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_class_init):
+ Initialize the GstRingerBuffer class to get it's debug category
+ initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
+ category and otherwise we get some g_critical(). Fixes bug #512334.
-2005-01-07 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/dv/demo-play.c: (main):
- xvideosink -> xvimagesink
- * ext/dv/gstdvdec.c:
- change rgb 32/32 caps to 24/32 (no alpha)
- change nb of channels to be a list (2 or 4, not 2)
- change sample rate to be a list (32, 44.1, 48 kHz) not a range
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- Add 'date/year' to extracted metadata list
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+=== release 0.10.16 ===
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
- The return value of fixate_to does not imply that the requested
- value was set, so don't assume.
+2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-07 Gergely Nagy <algernon@bonehunter.rulez.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ Release 0.10.16
+ Original commit message from CVS:
+ Release 0.10.16
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/libpng/gstpngdec.c:
- * ext/libpng/gstpngenc.c: (gst_pngenc_base_init),
- (gst_pngenc_sinklink), (gst_pngenc_init), (gst_pngenc_chain):
- * ext/libpng/gstpngenc.h:
- Alpha support (encoder; #163161), mime fixage.
+ * common:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
+ Original commit message from CVS:
+ Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_get_extension_data):
+ Fix typos and wrong extension check. Fixes #511274.
+
+2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
+ Original commit message from CVS:
+ * po/sk.po:
+ Oops - add new sk.po mentioned in the LINGUAS I just committed
+
+2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ po/LINGUAS: Add ca translation to the disted list.
+ Original commit message from CVS:
+ * po/LINGUAS:
+ Add ca translation to the disted list.
+ * win32/vs6/libgstsdp.dsp:
+ Convert line endings to CRLF
+
+2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ Add win32/vs6/libgstrtsp.dsp to MANIFEST
+
+2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Update for API changes in GIO and require GIO 2.15.2 for this.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/check/pipelines/gio.c: (GST_START_TEST):
+ Update for API changes in GIO and require GIO 2.15.2 for this.
+
+2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/common/: Add new API declarations
+ Original commit message from CVS:
+ * win32/common/libgstsdp.def:
+ * win32/common/libgstvideo.def:
+ Add new API declarations
+
+2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraparse.c:
+ Take a 2nd stab at handling libtheora granulepos changes in the decoder
+ and parser by inspecting the bitstream version of the incoming data.
+
+2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Provide one pkg-config file for every gst-plugins-base library.
+ Original commit message from CVS:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-audio-uninstalled.pc.in:
+ * pkgconfig/gstreamer-audio.pc.in:
+ * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
+ * pkgconfig/gstreamer-cdda.pc.in:
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft.pc.in:
+ * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast.pc.in:
+ * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
+ * pkgconfig/gstreamer-interfaces.pc.in:
+ * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
+ * pkgconfig/gstreamer-netbuffer.pc.in:
+ * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
+ * pkgconfig/gstreamer-pbutils.pc.in:
+ * pkgconfig/gstreamer-riff-uninstalled.pc.in:
+ * pkgconfig/gstreamer-riff.pc.in:
+ * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtp.pc.in:
+ * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp.pc.in:
+ * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
+ * pkgconfig/gstreamer-sdp.pc.in:
+ * pkgconfig/gstreamer-tag-uninstalled.pc.in:
+ * pkgconfig/gstreamer-tag.pc.in:
+ * pkgconfig/gstreamer-video-uninstalled.pc.in:
+ * pkgconfig/gstreamer-video.pc.in:
+ Provide one pkg-config file for every gst-plugins-base library.
+ This makes linking to those libraries much more intuitive and
+ provides standard pkg-config behaviour for them. Fixes bug #499697.
+
+2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
+
+ gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
+ Original commit message from CVS:
+ * gst/videoscale/vs_4tap.c:
+ Fix valgrind error on 4tap scaling method.
-2005-01-07 Sebastien Cote <sc5@hermes.usherb.ca>
+2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
+ Original commit message from CVS:
+ * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
+ Include Winsock2.h for VS6 and use a different way initialize
+ hints structure so it can build with VS6.
+ * win32/MANIFEST:
+ * win32/vs6/libgstsdp.dsp:
+ * win32/common/libgstsdp.def:
+ Add new files for libgstsdp.
+ * win32/vs6/grammar.dsp:
+ Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstdecodebin.dsp:
+ * win32/vs6/libgstdecodebin2.dsp:
+ * win32/vs6/libgstplaybin.dsp:
+ * win32/vs6/libgstvolume.dsp:
+ Add new dependencies to the link list.
+
+2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
+
+ win32/common/: Update/Add generated files in the win32 build directory.
+ Original commit message from CVS:
+ 2008-01-13 Julien Moutte <julien@fluendo.com>
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
+ (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
+ (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
+ (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
+ (gst_rtsp_header_field_get_type),
+ (gst_rtsp_status_code_get_type):
+ * win32/common/interfaces-enumtypes.c:
+ (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
+ (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
+ (gst_mixer_track_flags_get_type),
+ (gst_tuner_channel_flags_get_type):
+ * win32/common/multichannel-enumtypes.c:
+ (gst_audio_channel_position_get_type):
+ * win32/common/pbutils-enumtypes.c:
+ (gst_install_plugins_return_get_type):
+ * win32/common/pbutils-enumtypes.h: Update/Add generated files
+ in the win32 build directory.
+
+2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
+ * tests/check/elements/audiorate.c: (do_perfect_stream_test):
+ * tests/check/elements/playbin.c:
+ * tests/check/libs/mixer.c: (test_element_interface_supported),
+ (gst_implements_interface_init):
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Fix various assignment type mismatches.
+
+2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/rtsp/Makefile.am:
+ Add test to see if hstrerror is available or if we need libresolv
+ (Solaris) for it, then use it in libgstrtsp.
+
+2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/tag/Makefile.am: Fix include path order
+ Original commit message from CVS:
+ * gst-libs/gst/tag/Makefile.am:
+ Fix include path order
+
+2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst-libs/gst/pbutils/.gitignore:
+ Ignore more and make buildbot happy
+ Original commit message from CVS:
+ Ignore more and make buildbot happy
+
+2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ (gst_install_plugins_context_copy),
+ (gst_install_plugins_context_get_type):
+ * gst-libs/gst/pbutils/install-plugins.h:
+ Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
+ for bindings.
+
+2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (gst_theora_dec_class_init),
+ (_theora_granule_frame), (_theora_granule_start_time),
+ (theora_dec_sink_convert), (theora_dec_decode_buffer):
+ Adapt for post-alpha meaning of granulepos, when we
+ have a newer version of libtheora.
+ * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
+ (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
+ (theora_enc_is_discontinuous), (theora_enc_chain):
+ Likewise.
+ * tests/check/Makefile.am:
+ Link libtheora into theoraenc test so we can check which version of
+ libtheora we're testing against.
+ * tests/check/pipelines/theoraenc.c: (check_libtheora),
+ (check_buffer_granulepos),
+ (check_buffer_granulepos_from_starttime), (GST_START_TEST),
+ (theoraenc_suite):
+ Adapt tests to check the values that are now defined for theora; make
+ the tests backwards-adapt the passed values if we're running against an
+ old libtheora.
+ Fixes #497964
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faac/gstfaac.c: (gst_faac_outputformat_get_type),
- (gst_faac_class_init), (gst_faac_init), (gst_faac_srcconnect),
- (gst_faac_set_property), (gst_faac_get_property):
- * ext/faac/gstfaac.h:
- Allow for ADTS output (#153434).
+ gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ Ref audio clock class from a thread-safe context to make sure
+ we're not bit by GObjects lack of thread-safety here (#349410),
+ however unlikely that may be in practice.
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Fix against template (#150576).
+ autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
+ Original commit message from CVS:
+ * autogen.sh:
+ Add -Wno-portability to the automake parameters to stop warnings
+ about GNU make extensions being used. We require GNU make in almost
+ every Makefile anyway.
+ * configure.ac:
+ Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
+ at the same time is required for per target flags.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/games/gstpuzzle.c: (draw_puzzle):
- don't draw a puzzle if either width or height of tiles would be 0.
+ gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
+ Post an error message if we can't pull as many bytes as we need
+ for the tag. This makes sure the user gets to see a proper error
+ message if a file with a partial ID3 tag is fed to decodebin, and
+ not a 'no ID3 tag demuxer' error, which would be confusing
+ (see #508138).
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/games/gstpuzzle.c: (gst_puzzle_get_type),
- (gst_puzzle_class_init), (gst_puzzle_finalize):
- no memleaks, please
- (gst_puzzle_create), (gst_puzzle_init),
- (gst_puzzle_set_property), (gst_puzzle_setup):
- change initialization code around so we don't reshuffle on resize
- (draw_puzzle):
- fix another stupid typo
+ gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats):
+ Add description strings for ID3, APE, and ICY tags.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/games/gstvideoimage.c: (copy_hline_YUY2):
- fix stupid typo that borked copying on YUY2
+ gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1):
+ Make sure we error out correctly if we can't activate one of
+ the elements we've added. Fixes #508138.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
- * gst/games/gstpuzzle.c: (draw_puzzle):
- fix edges when image sizes aren't multiples of tile sizes
+ ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
+ Original commit message from CVS:
+ Patch by: Bastien Nocera <hadess at hadess net>
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
+ (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
+ Use snd_mixer_selem_set_{playback|capture}_volume_all() if
+ the volume is the same for all channels. This works around
+ some problem in alsa that leaves us with inconsistent state
+ for some reason (#486840).
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
- * gst/games/gstpuzzle.c: (gst_puzzle_base_init):
- make RGB endianness work correctly
- (gst_puzzle_show), (gst_puzzle_swap), (gst_puzzle_move):
- refactor and fix race with initial shuffling
- (nav_event_handler):
- allow using the mouse to puzzle
- (draw_puzzle):
- insist on tiles having width and height as multiples of 4 to get
- clean YUV image handling
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_buffer_alloc):
- s/DEBUG/LOG/ for common messages
- (gst_xvimagesink_navigation_send_event):
- fix mouse event translation to not include screen PAR
- * sys/ximage/ximagesink.c: (gst_ximagesink_navigation_send_event):
- fix mouse event translation to actually work
+ ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
+ Original commit message from CVS:
+ Patch by: Jerone Young <jerone at gmail com>
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
+ If there's no mixer track by the name of 'Master' or 'Front',
+ check if there's one called 'PCM' before trying the generic
+ fallback logic (fixes #506928, where we pick 'Mic' as master
+ track for the AD1984 card in a Thinkpad T61/X61 laptop).
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplay-enum.*: Add enums for configuration flags.
+ Original commit message from CVS:
+ * gst/playback/gstplay-enum.c:
+ (register_gst_autoplug_select_result),
+ (gst_autoplug_select_result_get_type), (register_gst_play_flags),
+ (gst_play_flags_get_type):
+ * gst/playback/gstplay-enum.h:
+ Add enums for configuration flags.
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
+ (gst_play_bin_get_property), (no_more_pads_cb),
+ (autoplug_select_cb), (gst_play_bin_change_state):
+ Merge mode with flags.
+ Add more property getters/setters, defaults and docs.
+ Add properties to get number of audio/video/text streams.
+ Create sink object in _init so that we can always rely on it being
+ there.
+ * gst/playback/gstplaysink.c: (gst_play_sink_init),
+ (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
+ (activate_vis), (gst_play_sink_reconfigure),
+ (gst_play_sink_set_flags), (gst_play_sink_get_flags),
+ (gst_play_sink_change_state):
+ * gst/playback/gstplaysink.h:
+ Use flags to configure the sink pipelines.
+ Add tee before audio pipeline so that we can use it for visualisations.
+ Start working on integrating visualisations.
+ Remove mode, we can do everything with the flags now.
+ Add method to configue the sink pipeline.
+
+2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
+ * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
+ Update to GMemoryInputStream API changes in GLib SVN and require
+ gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
+ We can also report the duration for every GSeekable, not only
+ GFileInputStream and GMemoryInputStream.
+
+2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
+ (check_buffer_timestamp), (check_buffer_duration):
+ Turn these functions into macros so we can see right away
+ where the failure occured.
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- Extract TrackNumber metadata + clean up code
- * gst/games/gstvideoimage.c: (gst_video_image_draw_rectangle):
- Hope this is the good fix (var used unitialised)
+2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
+ Original commit message from CVS:
+ 2008-01-05 Julien Moutte <julien@fluendo.com>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
+ debugging information to understand how X calculates the stride
+ for XvImages.
+
+2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
+ Original commit message from CVS:
+ * gst/volume/Makefile.am:
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
+ (volume_setup):
+ * gst/volume/gstvolume.h:
+ Use GstAudioFilter as base class for the volume element instead of
+ plain GstBaseTransform.
+
+2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
+ Don't set element details for the abstract GstAudioFilter class.
+
+2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
+ Implement get_unit_size() vmethod of GstBaseTransform.
+
+2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/pbutils.h:
+ Use glib-enum generator to have a proper enum GType for
+ GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
+
+2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
+
+ tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/theoraenc.c:
+ Reenable theoraenc test, which fails on the buildbot but
+ not locally.
-2005-01-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
- * ext/faad/gstfaad.c: (gst_faad_chain):
- Only increment timestamp if it's valid. Fixes raw AAC streams.
+ docs/: Add *-undeclared.txt to fix buildbot.
+ Original commit message from CVS:
+ * docs/libs/.cvsignore:
+ * docs/plugins/.cvsignore:
+ Add *-undeclared.txt to fix buildbot.
-2005-01-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
- * configure.ac:
- * gst/games/Makefile.am:
- * gst/games/gstpuzzle.c:
- add a puzzle game with...
- * gst/games/gstvideoimage.c:
- * gst/games/gstvideoimage.h:
- ... full colorspace support (that includes YUV9 and RGB16)) stolen
- from videotestsrc and made into something that would be a nice
- library for a lot of other plugins.
+ tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Second attempt at disabling theoraenc test long enough to
+ get buildbot to compile -base.
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
- * configure.ac:
- don't compile faad plugin if a RC of 2.0 is found
- Fixes #155346 (and FC1 buildbot)
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- try to make Solaris compiler happier
+ tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c:
+ Disable theoraenc test long enough to get the buildbot to
+ compile a recent -base.
-2005-01-06 Paul Jack <pjack@sfaf.org>
+2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (stop_cb):
+ Make sure we reset the slider value to 0.0 without racing against a
+ possible g_idle that sets it to something else.
+
+2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ sys/ximage/ximagesink.c: fix typo
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c:
+ fix typo
+
+2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ Add Location header so that we can start implementing redirects.
+ See #506025.
+
+2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ gst/subparse/gstssaparse.c: combine if's
+ Original commit message from CVS:
+ * gst/subparse/gstssaparse.c:
+ combine if's
+
+2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ gst/subparse/gstssaparse.c: remove duplicate log message
+ Original commit message from CVS:
+ * gst/subparse/gstssaparse.c:
+ remove duplicate log message
+
+2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gio/gstgio.c:
+ * ext/gio/gstgio.h:
+ * ext/gio/gstgiobasesink.h:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
+ * ext/gio/gstgiobasesrc.h:
+ * ext/gio/gstgiosink.c: (gst_gio_sink_start):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.h:
+ * ext/gio/gstgiostreamsink.h:
+ * ext/gio/gstgiostreamsrc.h:
+ * tests/check/pipelines/gio.c:
+ Update to latest API changes in GLib/GIO and require at least
+ gio-2.0 2.15.0 for this.
+ * ext/gio/Makefile.am:
+ Add GST_PLUGIN_LDFLAGS to LDFLAGS.
+
+2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_chain):
+ Fix 'xyz may be used uninitialized' compiler warnings caused
+ by broken g_assert_not_reached() macro in GLib-2.15.x and don't
+ abort() in any case but properly report the error.
+
+2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Code cleanups.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_finalize), (gst_play_bin_set_uri),
+ (gst_play_bin_set_suburi), (gst_play_bin_set_property),
+ (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
+ (autoplug_select_cb), (activate_group), (deactivate_group),
+ (setup_next_source), (save_current_group),
+ (gst_play_bin_change_state):
+ Code cleanups.
+ Remove next-uri, we can use the uri property just fine.
+ Fix some crasher.
+ Unref uridecodebin when switching.
+ Fix going to READY.
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gst_play_sink_init), (gst_play_sink_dispose),
+ (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
+ (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
+ (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
+ (gst_play_sink_set_property), (gst_play_sink_get_property),
+ (gen_video_chain), (gen_text_element), (gen_audio_chain),
+ (gen_vis_element), (gst_play_sink_get_mode),
+ (gst_play_sink_set_mode), (gst_play_sink_set_flags),
+ (gst_play_sink_get_flags), (gst_play_sink_request_pad),
+ (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
+ (gst_play_sink_change_state):
+ * gst/playback/gstplaysink.h:
+ Add some locking to make things threadsafe.
+ * gst/playback/test7.c: (about_to_finish_cb):
+ Fix test.
+
+2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
+ (gst_video_scale_get_property), (gst_video_scale_transform_caps),
+ (gst_video_scale_transform):
+ Don't claim to be able to handle/transform caps that can't really
+ be handled by the currently selected scaling method (here: RGB or
+ packed YUV with 4-tap method). Also add locking to method property.
+ * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
+ (test_basetransform_based):
+ Some test pipelines for the above (not entirely valgrind clean yet
+ apparently).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
- * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
- Fix segfault (#161667).
+ gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Add additional RGBA and RGB-24 video formats.
-2005-01-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Fix framerate reporting.
+ tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
+ Original commit message from CVS:
+ * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
+ (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
+ (test_suburi_error_wrongproto), (test_missing_primary_decoder):
+ * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
+ (cddabasesrc_suite):
+ Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
+ deprecated in the future (see #498924).
-2005-01-05 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-ids.h:
- * gst/wavenc/riff.h:
- Add AMR (VBR and CBR) ids to riff.h audio codec list
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc),
- (gst_asf_demux_process_object):
- Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
-
-2005-01-05 Martin Eikermann <meiker@upb.de>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
- (gst_dvd_demux_handle_discont):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_class_init),
- (gst_mpeg_demux_handle_discont):
- Recreate pads on new-media (#160730).
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_new_pad):
- Send discont even if manager changes timestamps (#161929).
-
-2005-01-05 Sebastien Cote <sc5@hermes.usherb.ca>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16):
- Fix invalid memory access (#159211).
-
-2005-01-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * examples/gstplay/player.c: (main):
- Don't iterate.
- * examples/seeking/seek.c: (fixate), (make_playerbin_pipeline):
- Add visualizations.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_push),
- (gst_a52dec_handle_frame):
- Set duration.
- * ext/dvdnav/gst-dvd:
- Add audioconvert. Fixes #161325.
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get):
- Explicitely case to gint64. Possible valgrind error.
- * gst-libs/gst/play/play.c: (caps_set), (setup_size),
- (gst_play_tick_callback), (gst_play_change_state),
- (gst_play_dispose), (gst_play_init), (gst_play_class_init),
- (gst_play_set_location), (gst_play_get_location),
- (gst_play_seek_to_time), (gst_play_set_data_src),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization), (gst_play_connect_visualization),
- (gst_play_get_framerate), (gst_play_get_all_by_interface),
- (gst_play_new):
- Use playbin. Fixes #139749 and #147744.
- * gst/apetag/apedemux.c: (gst_ape_demux_parse_tags):
- Add genre tag.
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
- (audioscale_get_type), (gst_audioscale_base_init),
- (gst_audioscale_class_init), (gst_audioscale_expand_caps),
- (gst_audioscale_getcaps), (gst_audioscale_fixate),
- (gst_audioscale_link), (gst_audioscale_get_buffer),
- (gst_audioscale_decrease_rate), (gst_audioscale_increase_rate),
- (gst_audioscale_init), (gst_audioscale_dispose),
- (gst_audioscale_chain), (gst_audioscale_set_property),
- (gst_audioscale_get_property), (plugin_init):
- Indent properly.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private):
- Fix LPCM.
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta),
- (qtdemux_tag_add_str), (qtdemux_tag_add_num),
- (qtdemux_tag_add_gnre), (qtdemux_video_caps):
- Add more metadata (fixes #162656).
-
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gststreamselector.c: Don't leak event.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_selector_pad_event):
+ Don't leak event.
- * configure.ac:
- back to cvs
+2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
-=== release 0.8.7 ===
+ gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c:
+ Use GST_ROUND_UP_2 macro
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.7, "Hyperspace"
+ gst/playback/.cvsignore: Ignore more.
+ Original commit message from CVS:
+ * gst/playback/.cvsignore:
+ Ignore more.
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
+ Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
+ * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
+ (set_active_source):
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (setup_sinks), (playbin_set_subtitles_visible):
+ Make switching off of subtitles work. To avoid all kind of
+ problems with unlinking of the subtitle input, we just keep
+ the subtitle inputs linked as they are and tell textoverlay
+ not to render them. Fixes #373011.
+ Other subtitle switching issues (esp. when there are both
+ external and in-stream subtitles) remain. They'll be solved
+ in playbin2.
- * gst/playback/gstplaybasebin.c:
- Fix for #162924 - free caps after use, not before
+2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gststreamselector.c: Init the pad segment too.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_selector_pad_init):
+ Init the pad segment too.
+
+2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
+ (gst_audioringbuffer_open_device),
+ (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
+ (gst_audioringbuffer_release), (gst_audioringbuffer_start),
+ (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
+ (gst_audio_sink_create_ringbuffer):
+ Improve debug output.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (gst_ring_buffer_delay):
+ Prevent some functions from doing things and failing when the
+ ringbuffer is not yet acquired.
+
+2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/interfaces.h:
+ Also remove interfaces.h from CVS as it is not needed anymore.
-2005-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/Makefile.am:
+ interfaces.h is not used anymore so remove it from the build
+ process.
- * gst/playback/gstplaybasebin.c:
- * gst/wavparse/gstwavparse.c:
- Fix for #154773 - fixes playback of small .wav files
+2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
-2005-01-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Add a "blink" pattern. Turn on the pain. Apologies. It's useful
+ for testing vertical refresh synchronization.
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
- * gst/audioscale/gstaudioscale.c:
- Fix for #162819 - make audioscale reusable
- Fixes playback of more than one file with playbin/totem
+ Add new GstVideFormat enum and write a bunch of helper functions based around it.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Add new GstVideFormat enum and write a bunch of helper functions
+ based around it.
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- * gst/ffmpegcolorspace/imgconvert.c:
- clean up the mess that made me cry and avoid needless duplication
+ Makefile.am: Use new common/win32.mak.
+ Original commit message from CVS:
+ * Makefile.am:
+ Use new common/win32.mak.
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create), (gst_base_audio_src_change_state):
+ Add debug info.
+ When going from PLAYING to PAUSED, pause the ringbuffer before calling
+ the parent state change function, just like the audiosink, because the
+ parent waits for the element to finish its processing before completing
+ the state change. This makes going to PAUSED a lot snappier.
+ When going from READY to PAUSED, don't allow the ringbuffer to start
+ yet.
+
+2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Yet another fix for broken software that produce files with an empty
+ blockalign field. Instead of completely failing, make a second attempt
+ at guessing the width/depth by looking at strf->size.
+
+2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
+ (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
+ * gst-libs/gst/pbutils/install-plugins.c:
+ (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ (gst_missing_plugin_message_get_installer_detail),
+ (gst_missing_encoder_installer_detail_new):
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
+ Turn a few g_assert_not_reached() into g_return_val_if_reached() to
+ avoid compiler warnings (#503930).
- * gst/ffmpegcolorspace/imgconvert.c:
- give some indication of why we're segfaulting
+2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-12-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
+ for jpeg video streams.
+ Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
+ for the above modification.
- * configure.ac:
- Fix indentation, fix v4l2 plugin detection.
- * ext/Makefile.am:
- Fix libmms location (Maciej, use diff -u!).
- * ext/alsa/gstalsa.c: (gst_alsa_init):
- Initialize caps cache to NULL.
- * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
- Only change state on audiosink if it exists.
-
-2004-12-28 Maciej Katafiasz <mathrick@mathrick.org>
-
- * gst/matroska/matroska-demux.c:
- * gst/matroska/matroska-ids.h:
- * gst/matroska/matroska-demux.h:
- Fix Vorbis streams failing to decode in some files, where cluster_time
- isn't 0, because then it doesn't send codec_priv before actual data.
- Remove time-based test and replace it with marker set on beginning of
- new stream
-
-2004-12-28 David Schleef <ds@schleef.org>
-
- Merge patch from Ronald fixing problems with streaming
- text.
- * ext/cairo/gstcairo.c: (plugin_init):
- * ext/cairo/gsttextoverlay.c: (gst_textoverlay_render_text),
- (gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
- (gst_textoverlay_video_chain), (gst_textoverlay_loop),
- (gst_textoverlay_font_init), (gst_textoverlay_init),
- (gst_textoverlay_set_property):
- * ext/cairo/gsttextoverlay.h:
-
-2004-12-27 David Schleef <ds@schleef.org>
-
- * ext/cairo/gsttextoverlay.c: (gst_textoverlay_render_text),
- (gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
- (gst_textoverlay_video_chain), (gst_textoverlay_loop),
- (gst_textoverlay_font_init), (gst_textoverlay_init),
- (gst_textoverlay_set_property): Improvements to actually
- render text as white on black outline on video, including
- font selection and horizontal/vertical alignment. (Ronald's
- christmas present)
- * ext/cairo/gsttextoverlay.h:
-
-2004-12-26 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstogg.c:
- * ext/ogg/gstogmparse.c:
- fix ogm[audio/video]parse plugin registration
- (riff won't load if bytestream is already loaded)
+ gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
+ (gst_x_overlay_handle_events):
+ More guards (we don't want klass to end up being NULL).
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/audioconvert/gstchannelmix.c:
- fix for GLIB < 2.4
+ Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/volume/gstvolume.c: (gst_volume_init):
+ Use new gst_base_transform_set_gap_aware() function as volume
+ correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
+ for this.
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
- * Makefile.am:
- * configure.ac:
- disable docs again until it actually passes make distcheck.
+ tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (msg_segment_done), (main):
+ Don't go to READY on EOS as this avoids testing of seeking and
+ restarting after EOS, use the stop button when you want to READY.
+ Don't try to do a flushing seek in segment-done, it does not make
+ sense to use this for gapless playback and is not needed.
-2004-12-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/qtdemux/qtdemux.c: (qtdemux_type_get), (qtdemux_audio_caps):
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (plugin_init):
- Add 3GP (variables name Q3GP because they can't start with a
- number). Add samr audio fourcc (used in .3gp files), decoder
- is work in progress. Also do a GST_WARNING instead of ERROR
- in case of unknown nodes, to decrease output.
+ gst/playback/gstqueue2.c: Use separate timers for input and output rates.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
+ (reset_rate_timer), (update_in_rates), (update_out_rates),
+ (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
+ (gst_queue_chain), (gst_queue_loop):
+ Use separate timers for input and output rates.
+ Pause measuring the output rate when we block for more data.
+ See #503262.
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
- * Makefile.am:
- really fix dist
+ * gst/speexresample/Makefile.am:
+ update spec file and add two missing files for disting
+ Original commit message from CVS:
+ update spec file and add two missing files for disting
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- * ext/speex/gstspeexdec.h:
- * ext/speex/gstspeexenc.h:
- Fixes #158382. Make speex plugin compatible with both 1.0 and 1.1.
- Fix detection code in configure.ac
+ gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_chain):
+ Pause the timer to measure the input rate when we block because the
+ queue is filled. See #503262.
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Save position, so that queries give proper return values. Don't
- know how this could ever have worked before...
+ gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_free):
+ Close control sockets. Fixes #503440.
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- Put additional LAME check inside the conditional. Fixes #152339
+ gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
+ Expose the right pad in the right place with the right element.
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan):
- Add some more debug. Fix logic error when setting movi offset
- while reading index.
+ gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats):
+ Add description for 'private' dts caps (who come up with that name?).
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan), (gst_avi_demux_handle_seek),
- (gst_avi_demux_process_next_entry):
- Add some debugging. Better detection of broken indexes and the
- accompanying index recovery. No infinite loops on state changes
- when we're still in our loopfunction.
+ Makefile.am: Add check-exports target and run it with 'make check'.
+ Original commit message from CVS:
+ * Makefile.am:
+ Add check-exports target and run it with 'make check'.
+ * configure.ac:
+ Be stricter about what we export in our libraries: change regexp so that
+ we only export _gst_foo(), but not __gst_foo().
+ * gst-libs/gst/cdda/base64.h: (rfc822_binary):
+ * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
+ Change internal functions to __gst_foo so they dont' get exported.
+ * win32/common/libgstaudio.def:
+ Add missing symbols.
-2004-12-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
- * configure.ac:
- Fix up.
+ * ChangeLog:
+ ChangeLog: remove conflict markers
+ Original commit message from CVS:
+ ChangeLog: remove conflict markers
-2004-12-22 Archana Shah <archana.shah@wipro.com>
+2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
+ Original commit message from CVS:
+ * ext/gnomevfs/Makefile.am:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
+ Use gst_tag_freeform_string_to_utf8() here, which also takes
+ into account any character sets specified by the user via
+ environment variables.
- * sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_volume):
- Normalizing the value before setting
- (gst_sunaudiomixer_get_volume):
- Normalizing the value after getting. Fixes bug# 161980
+2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-22 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ gst/audioconvert/Makefile.am: Also link to libm.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Also link to libm.
- * Makefile.am: Make sure docs gets disted
- * docs/Makefile.am: Make sure all needed files get disted
- * gst-plugins.spec.in: latest updates
+2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-22 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ No need for floating point operations here. avoids having to link
+ against the math library too.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- Revert patch 1.38 as clock distribution over schedulers does
- not work correcly in the core yet.
+2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats),
+ (format_info_get_desc):
+ * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
+ (GST_START_TEST):
+ Add one or two missing formats. Generate ADPCM description
+ dynamically depending on layout/format.
- * sys/oss/README: remove this file, which predates my birth
- (and which content is by far outdated)
+2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-12-20 Stefan Kost <ensonic@users.sf.net>
+ configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
+ Original commit message from CVS:
+ * configure.ac:
+ Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
- * Makefile.am:
- * configure.ac:
- * docs/Makefile.am:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-libs-docs.sgml:
- * docs/libs/gst-plugins-libs-sections.txt:
- * docs/libs/tmpl/gstgconf.sgml:
- * docs/upload.mak:
- * docs/version.entities.in:
- Added boilerplate gtk-doc files for plugin-libs documentation.
+2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
-2004-12-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
+ Original commit message from CVS:
+ Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
+ Some .srt files start with chunk number 0 and not chunk number 1,
+ recognise and accept those as well (fixes #502497).
+ * tests/check/elements/subparse.c: (srt_input), (srt_input0),
+ (test_src):
+ Add unit test for the above.
- * gst/auparse/gstauparse.c: fix int and float audio caps
+2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplay-enum.*: Add missing files.
+ Original commit message from CVS:
+ * gst/playback/gstplay-enum.c:
+ (register_gst_autoplug_select_result),
+ (gst_autoplug_select_result_get_type):
+ * gst/playback/gstplay-enum.h:
+ Add missing files.
+
+2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ Group decodebin2 and uridecodebin into the same plugin so that they
+ can share the GEnumType.
+ * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
+ (_gst_select_accumulator), (gst_decode_bin_class_init),
+ (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
+ (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
+ (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
+ Add signal to sort factories instead of the more awkward autoplug-select
+ signal.
+ Modify autoplug_select so that we can try, skip or expose the
+ autopluggin of an element on a pad.
+ * gst/playback/gstfactorylists.c: (compare_ranks),
+ (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
+ (element_filter), (gst_factory_list_get_elements),
+ (gst_factory_list_debug), (gst_factory_list_filter):
+ * gst/playback/gstfactorylists.h:
+ Simplify the API, allow getting elements based on mask.
+ * gst/playback/gstplay-marshal.list:
+ Add some more marshallers.
+ * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
+ (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
+ (autoplug_select_cb), (activate_group):
+ Add support for managing non-raw sinks by providing a custom element and
+ sink list to decodebin2.
+ Try to plug non-raw sinks when decodebin2 using autoplug-select of
+ decodebin2.
+ * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
+ (gst_play_sink_set_mode), (gst_play_sink_request_pad):
+ * gst/playback/gstplaysink.h:
+ Add support for raw and non-raw sinks.
+ Add support to force sinks selected by playbin2.
+ Don't plug raw converters for non-raw sinks.
+ * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
+ (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
+ (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
+ (plugin_init):
+ Use right accumulators.
+ Proxy new signal.
+
+2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ Use runnning time as the base time instead of the timestamp.
+ Spotted by Saur on IRC.
+
+2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
+
+2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
+ (gst_ogg_demux_read_chain):
+ If we find a new serial number but it does not contain a BOS page, make
+ sure we initialize the chain to NULL because else we will try to scan it
+ and crash. Fixes #500763
+
+2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Refactor some common code to filter factories and check caps compat.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
+ (get_feature_array), (decoders_filter), (sinks_filter),
+ (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
+ (gst_factory_list_filter):
+ * gst/playback/gstfactorylists.h:
+ Refactor some common code to filter factories and check caps compat.
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_init), (gst_decode_bin_dispose),
+ (gst_decode_bin_autoplug_continue),
+ (gst_decode_bin_autoplug_factories),
+ (gst_decode_bin_autoplug_select), (analyze_new_pad),
+ (find_compatibles):
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_init), (gst_play_bin_finalize),
+ (autoplug_factories_cb), (activate_group):
+ * gst/playback/gstqueue2.c:
+ * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
+ (proxy_autoplug_continue_signal),
+ (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
+ (proxy_drained_signal):
+ Add some more debug info and use factor filtering code.
+
+2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
+
+ configure.ac: Add QuickTime Wrapper plug-in.
+ Original commit message from CVS:
+ 2007-11-26 Julien Moutte <julien@fluendo.com>
+ * configure.ac: Add QuickTime Wrapper plug-in.
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
+ build on Mac OS X Leopard. Incorrect printf format arguments.
+ * sys/Makefile.am:
+ * sys/qtwrapper/Makefile.am:
+ * sys/qtwrapper/audiodecoders.c:
+ (qtwrapper_audio_decoder_base_init),
+ (qtwrapper_audio_decoder_class_init),
+ (qtwrapper_audio_decoder_init),
+ (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
+ (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
+ (make_samr_magic_cookie), (open_decoder),
+ (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
+ (qtwrapper_audio_decoder_chain),
+ (qtwrapper_audio_decoder_sink_event),
+ (qtwrapper_audio_decoders_register):
+ * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
+ (fourcc_to_caps):
+ * sys/qtwrapper/codecmapping.h:
+ * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
+ (image_description_for_mp4v), (image_description_from_stsd_buffer),
+ (image_description_from_codec_data):
+ * sys/qtwrapper/imagedescription.h:
+ * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
+ (get_output_info_from_component), (dump_avcc_atom),
+ (dump_image_description), (dump_codec_decompress_params),
+ (addSInt32ToDictionary), (dump_cvpixel_buffer),
+ (DestroyAudioBufferList), (AllocateAudioBufferList):
+ * sys/qtwrapper/qtutils.h:
+ * sys/qtwrapper/qtwrapper.c: (plugin_init):
+ * sys/qtwrapper/qtwrapper.h:
+ * sys/qtwrapper/videodecoders.c:
+ (qtwrapper_video_decoder_base_init),
+ (qtwrapper_video_decoder_class_init),
+ (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
+ (fill_image_description), (new_image_description), (close_decoder),
+ (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
+ (decompressCb), (qtwrapper_video_decoder_chain),
+ (qtwrapper_video_decoder_sink_event),
+ (qtwrapper_video_decoders_register): Initial import of QuickTime
+ wrapper jointly developped by Songbird authors (Pioneers of the
+ Inevitable) and Fluendo.
+
+2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/: Add GAP-flag support.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ Add GAP-flag support.
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/resample.c: (resampler_basic_direct_single),
+ (resampler_basic_direct_double),
+ (resampler_basic_interpolate_single),
+ (resampler_basic_interpolate_double),
+ (speex_resampler_process_native), (speex_resampler_process_float),
+ (speex_resampler_process_int),
+ (speex_resampler_process_interleaved_float),
+ (speex_resampler_process_interleaved_int),
+ (speex_resampler_get_input_latency),
+ (speex_resampler_get_output_latency):
+ * gst/speexresample/speex_resampler.h:
+ Update speex resampler to latest SVN. We're now down to only the
+ changes noted in README again.
+ * gst/speexresample/speex_resampler_wrapper.h:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_push_drain), (gst_speex_resample_query):
+ Adjust to API changes.
+
+2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
+ Original commit message from CVS:
+ 2007-11-24 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (main): Increase the range of the
+ rate selector as I would like to test QOS behavior at higher
+ forward and reverse playback speed like say 64x.
+
+2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_update_state):
+ Only post the latency message if we have a resampler state already.
+
+2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioresample/gstaudioresample.c: Implement latency query.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
+ (audioresample_query), (audioresample_query_type),
+ (gst_audioresample_set_property):
+ Implement latency query.
+
+2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_update_state):
+ Also post GST_MESSAGE_LATENCY if the latency changes.
+
+2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
+ Original commit message from CVS:
+ * gst/speexresample/resample.c: (speex_resampler_get_latency),
+ (speex_resampler_drain_float), (speex_resampler_drain_int),
+ (speex_resampler_drain_interleaved_float),
+ (speex_resampler_drain_interleaved_int):
+ * gst/speexresample/speex_resampler.h:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ Add functions to push the remaining samples and to get the latency
+ of the resampler. These will get added to Speex SVN in this or a
+ slightly changed form at some point too and should get merged then
+ again.
+ * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
+ (gst_speex_resample_init_state),
+ (gst_speex_resample_transform_size),
+ (gst_speex_resample_push_drain), (gst_speex_resample_event),
+ (gst_speex_fix_output_buffer), (gst_speex_resample_process),
+ (gst_speex_resample_query), (gst_speex_resample_query_type):
+ Drop the prepending zeroes and output the remaining samples on EOS.
+ Also properly implement the latency query for this. speexresample
+ should be completely ready for production use now.
+
+2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain):
+ Our EOS time contains the base_time, _wait_eos() expects a running_time
+ so we have to subtract the base_time again before calling the function.
+ This fixes an EOS regression where the base_time was added twice and EOS
+ took longer and longer in certain situations.
+ Fixes #498767.
+
+2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Expose methods for some object properties so that subclasses can more easily configure them.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
+ (gst_base_audio_sink_set_provide_clock),
+ (gst_base_audio_sink_get_provide_clock),
+ (gst_base_audio_sink_set_slave_method),
+ (gst_base_audio_sink_get_slave_method),
+ (gst_base_audio_sink_set_property),
+ (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
+ (gst_base_audio_sink_none_slaving),
+ (gst_base_audio_sink_handle_slaving):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Expose methods for some object properties so that subclasses can more
+ easily configure them.
+ Added slave method none, that completely disables slaving to the
+ internal clock.
+ API: gst_base_audio_sink_set_provide_clock()
+ API: gst_base_audio_sink_get_provide_clock()
+ API: gst_base_audio_sink_set_slave_method()
+ API: gst_base_audio_sink_get_slave_method()
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_set_provide_clock),
+ (gst_base_audio_src_get_provide_clock),
+ (gst_base_audio_src_set_property),
+ (gst_base_audio_src_get_property), (gst_base_audio_src_create):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ Expose methods for some object properties so that subclasses can more
+ easily configure them.
+ API: gst_base_audio_src_set_provide_clock()
+ API: gst_base_audio_src_get_provide_clock()
+
+2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ Add README explaining where the resampling code was taken from
+ and which changes were done.
+ * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
+ (speex_free):
+ Use g_malloc() and friends instead of malloc() to achieve higher
+ portability and define the functions inline.
+ * gst/speexresample/speex_resampler.h:
+ Add back some useless preprocessor stuff to keep the diff between
+ our version and the one from the Speex SVN repository lower.
+
+2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
+ Some small cleanup and addition of a TODO item.
+
+2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/Makefile.am: Add missing file.
+ Original commit message from CVS:
+ * gst/speexresample/Makefile.am:
+ Add missing file.
+
+2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
+ Original commit message from CVS:
+ Patch by: Joe Peterson <lavajoe at gentoo dot org>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Fix compilation on FreeBSD (Gentoo). Fixes #498228.
+
+2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add speexresample to the docs and while at that do a make update.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * docs/plugins/gst-plugins-bad-plugins.args:
+ * docs/plugins/gst-plugins-bad-plugins.signals:
+ * docs/plugins/inspect/plugin-bz2.xml:
+ * docs/plugins/inspect/plugin-cdxaparse.xml:
+ * docs/plugins/inspect/plugin-dtsdec.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-faac.xml:
+ * docs/plugins/inspect/plugin-faad.xml:
+ * docs/plugins/inspect/plugin-filter.xml:
+ * docs/plugins/inspect/plugin-freeze.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gsm.xml:
+ * docs/plugins/inspect/plugin-gstrtpmanager.xml:
+ * docs/plugins/inspect/plugin-h264parse.xml:
+ * docs/plugins/inspect/plugin-modplug.xml:
+ * docs/plugins/inspect/plugin-mpeg2enc.xml:
+ * docs/plugins/inspect/plugin-musepack.xml:
+ * docs/plugins/inspect/plugin-musicbrainz.xml:
+ * docs/plugins/inspect/plugin-nsfdec.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-soundtouch.xml:
+ * docs/plugins/inspect/plugin-spcdec.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speed.xml:
+ * docs/plugins/inspect/plugin-tta.xml:
+ * docs/plugins/inspect/plugin-videosignal.xml:
+ * docs/plugins/inspect/plugin-xingheader.xml:
+ * docs/plugins/inspect/plugin-xvid.xml:
+ * gst/speexresample/gstspeexresample.h:
+ Add speexresample to the docs and while at that do a make update.
+
+2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_fix_output_buffer), (gst_speex_resample_process):
+ If the resampler gives less output samples than expected
+ adjust the output buffer and print a warning.
+
+2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add resample element based on the Speex resampling algorithm.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/fixed_generic.h:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_base_init), (gst_speex_resample_class_init),
+ (gst_speex_resample_init), (gst_speex_resample_start),
+ (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
+ (gst_speex_resample_transform_caps),
+ (gst_speex_resample_init_state), (gst_speex_resample_update_state),
+ (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
+ (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
+ (gst_speex_resample_event), (gst_speex_resample_check_discont),
+ (gst_speex_resample_process), (gst_speex_resample_transform),
+ (gst_speex_resample_set_property),
+ (gst_speex_resample_get_property), (plugin_init):
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
+ (speex_free), (compute_func), (main), (sinc), (cubic_coef),
+ (resampler_basic_direct_single), (resampler_basic_direct_double),
+ (resampler_basic_interpolate_single),
+ (resampler_basic_interpolate_double), (update_filter),
+ (speex_resampler_init), (speex_resampler_init_frac),
+ (speex_resampler_destroy), (speex_resampler_process_native),
+ (speex_resampler_process_float), (speex_resampler_process_int),
+ (speex_resampler_process_interleaved_float),
+ (speex_resampler_process_interleaved_int),
+ (speex_resampler_set_rate), (speex_resampler_get_rate),
+ (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
+ (speex_resampler_set_quality), (speex_resampler_get_quality),
+ (speex_resampler_set_input_stride),
+ (speex_resampler_get_input_stride),
+ (speex_resampler_set_output_stride),
+ (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
+ (speex_resampler_reset_mem), (speex_resampler_strerror):
+ * gst/speexresample/speex_resampler.h:
+ * gst/speexresample/speex_resampler_float.c:
+ * gst/speexresample/speex_resampler_int.c:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ Add resample element based on the Speex resampling algorithm.
+
+2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
+ Original commit message from CVS:
+ * tests/check/libs/fft.c: (GST_START_TEST):
+ Fix scaling to really have dB instead of something else.
+
+2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: There's a nice macro to check
+ Original commit message from CVS:
+ 2007-11-19 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (main): There's a nice macro to
+ check
+ GTK version, use it.
+
+2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Try to support stable version of GTK.
+ Original commit message from CVS:
+ 2007-11-19 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (main): Try to support stable version
+ of GTK.
+
+2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: Fix the build + little README update.
+ Original commit message from CVS:
+ * gst/playback/README:
+ * gst/playback/test7.c:
+ Fix the build + little README update.
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- * sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
- g_assert() can be a macro, don't use #ifdef inside it.
+2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add playbin2 seek pipeline.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
+ Add playbin2 seek pipeline.
+
+2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Add playbin2.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.c: (plugin_init):
+ * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
+ (eos_cb), (about_to_finish_cb), (main):
+ Add playbin2.
+ Added gapless playback example.
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
+ * gst/playback/gstqueue2.c:
+ * gst/playback/test.c:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
+ (pad_removed_cb):
+ * gst/playback/gststreaminfo.h:
+ Change email.
+ * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
+ (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
+ (gst_play_bin_dispose), (gst_play_bin_set_uri),
+ (gst_play_bin_set_suburi), (gst_play_bin_set_property),
+ (gst_play_bin_get_property), (gst_play_bin_handle_message),
+ (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
+ (drained_cb), (unlink_group), (activate_group),
+ (setup_next_source), (gst_play_bin_change_state),
+ (gst_play_bin2_plugin_init):
+ Added raw first version of playbin2. Does chained oggs and gapless
+ playback fine. No support for raw sinks yet. No visualisations or
+ subtitles yet.
+ * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
+ (gst_play_sink_class_init), (gst_play_sink_init),
+ (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
+ (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
+ (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
+ (gst_play_sink_set_property), (gst_play_sink_get_property),
+ (post_missing_element_message), (free_chain), (add_chain),
+ (activate_chain), (gen_video_chain), (gen_text_element),
+ (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
+ (gst_play_sink_set_mode), (gst_play_sink_request_pad),
+ (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
+ (gst_play_sink_send_event), (gst_play_sink_change_state):
+ * gst/playback/gstplaysink.h:
+ Added Element that abstracts the sinks and their pipelines for playbin2.
+
+2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
+ (gst_selector_pad_class_init), (gst_selector_pad_init),
+ (gst_selector_pad_finalize), (gst_selector_pad_reset),
+ (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
+ (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
+ (gst_selector_pad_chain), (gst_stream_selector_get_type),
+ (gst_stream_selector_base_init), (gst_stream_selector_class_init),
+ (gst_stream_selector_init), (gst_stream_selector_set_property),
+ (gst_stream_selector_get_linked_pad),
+ (gst_stream_selector_getcaps),
+ (gst_stream_selector_is_active_sinkpad),
+ (gst_stream_selector_activate_sinkpad),
+ (gst_stream_selector_get_linked_pads),
+ (gst_stream_selector_request_new_pad),
+ (gst_stream_selector_release_pad):
+ * gst/playback/gststreamselector.h:
+ Improve streamselector, make it select and unselect the current pad more
+ intelligently.
+ Subclass GstPad for the sinkpads of the selector.
+ Handle segments more correctly.
+ Fix caps negotiation.
+ Implement release_pad.
+
+2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_group_check_if_drained), (source_pad_event_probe),
+ (remove_fakesink):
+ Add drained signal fired when decodebin finishes decoding the data.
+ Remove deprecated STATE_DIRTY message.
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
+ (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
+ (analyse_source), (proxy_drained_signal), (make_decoder),
+ (source_new_pad), (value_list_append_structure_list),
+ (handle_redirect_message), (handle_message):
+ Proxy the new drained signal.
+ Handle pad removed from decodebin.
+ Handle redirect messages by sorting multiple redirections based on the
+ connection speed.
+
+2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
+
+ gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
+ Fix leaking headers. Fixes #496761.
+
+2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
+
+ sys/: Don't leak the PAR on errors. Fixes #496731.
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
+ (gst_ximagesink_change_state):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
+ Don't leak the PAR on errors. Fixes #496731.
+
+2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
+ (gst_tag_from_id3_user_tag):
+ Add mapping for audio cd discid tags, so we can extract
+ them from tags as well (see #347848). Also compare identifiers
+ in ID3v2 TXXX frames in a case-insensitive way to increase
+ compatibility when reading tags (discid vs. DiscID vs. DiscId).
-2004-12-19 Edward Hervey <bilboed@bilboed.com>
+2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-plugins-base.doap: Oops, fix the release name.
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Oops, fix the release name.
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_change_state):
- Event handling (fixes #159986).
+2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-plugins-base.doap: Add 0.10.15 release
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Add 0.10.15 release
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add BLZ0 (Blizzard's version of DivX) fourcc.
+2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-18 David Schleef <ds@schleef.org>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * gst/tta/gstttadec.c: (gst_tta_dec_link): And yet another
- portability fix.
+=== release 0.10.15 ===
-2004-12-18 David Schleef <ds@schleef.org>
+2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tta/ttadec.h: Disable some header code that isn't used
- and clearly isn't portable.
+ configure.ac: releasing 0.10.15, "No need to argue"
+ Original commit message from CVS:
+ === release 0.10.15 ===
+ 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+ * configure.ac:
+ releasing 0.10.15, "No need to argue"
-2004-12-18 David Schleef <ds@schleef.org>
+2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/ffmpegcolorspace/imgconvert.c: (get_pix_fmt_info),
- (avcodec_get_chroma_sub_sample), (avcodec_get_pix_fmt_name),
- (avcodec_get_pix_fmt), (avpicture_layout),
- (avcodec_get_pix_fmt_loss), (avg_bits_per_pixel), (img_copy),
- (get_convert_table_entry), (img_convert), (img_get_alpha_info):
- Fix code to not use GCC extensions (and c99 extensions that
- Forte does not like.)
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/vs6/libgstfft.dsp: Convert line endings to DOS.
+ Original commit message from CVS:
+ * win32/vs6/libgstfft.dsp:
+ Convert line endings to DOS.
+
+2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
+ Original commit message from CVS:
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstfft.dsp:
+ * win32/MANIFEST:
+ Add a project file for fft plugin and remove socket
+ based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgsttag.dsp:
+ Convert line endings back to DOS.
+ Fixes #496724
+
+2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/vs6/: Convert line endings back to DOS
+ Original commit message from CVS:
+ * win32/vs6/libgstinterfaces.dsp:
+ * win32/vs6/libgstrtsp.dsp:
+ Convert line endings back to DOS
+
+2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
+ Original commit message from CVS:
+ * gst-libs/gst/fft/kiss_fft_f32.h:
+ * gst-libs/gst/fft/kiss_fft_f64.h:
+ * gst-libs/gst/fft/kiss_fft_s16.h:
+ * gst-libs/gst/fft/kiss_fft_s32.h:
+ Don't include malloc.h which doesn't exist on Mac OSX.
+ Instead, pull in glib.h and use g_malloc/g_free for
+ consistency. Fixes: #496548
+
+2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Dont leak ghostpad. Fixes #475451.
-2004-12-19 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Update some more docs and comments.
+ Original commit message from CVS:
+ * docs/design/design-decodebin.txt:
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad):
+ Update some more docs and comments.
+
+2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Require GIO >= 0.1.2 and adjust unit test for an API change.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/check/pipelines/gio.c: (GST_START_TEST):
+ Require GIO >= 0.1.2 and adjust unit test for an API change.
+
+2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
+ Original commit message from CVS:
+ * ext/gio/gstgio.h:
+ Add macro to check if a stream supports seeking.
+ * ext/gio/Makefile.am:
+ * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
+ (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
+ (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
+ (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
+ (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
+ (gst_gio_base_sink_render), (gst_gio_base_sink_query),
+ (gst_gio_base_sink_set_stream):
+ * ext/gio/gstgiobasesink.h:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
+ (gst_gio_base_src_class_init), (gst_gio_base_src_init),
+ (gst_gio_base_src_finalize), (gst_gio_base_src_start),
+ (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
+ (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
+ (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
+ (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
+ * ext/gio/gstgiobasesrc.h:
+ Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
+ base classes that only require a GInputStream or GOutputStream to
+ work.
+ * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
+ (gst_gio_sink_class_init), (gst_gio_sink_init),
+ (gst_gio_sink_finalize), (gst_gio_sink_start):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
+ (gst_gio_src_class_init), (gst_gio_src_init),
+ (gst_gio_src_finalize), (gst_gio_src_start):
+ * ext/gio/gstgiosrc.h:
+ Use the newly created base classes here.
+ * ext/gio/gstgio.c: (plugin_init):
+ * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
+ (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
+ (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
+ (gst_gio_stream_sink_get_property):
+ * ext/gio/gstgiostreamsink.h:
+ * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
+ (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
+ (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
+ (gst_gio_stream_src_get_property):
+ * ext/gio/gstgiostreamsrc.h:
+ Implement GstGioStreamSink and GstGioStreamSrc that have a property
+ to set the GInputStream/GOutputStream that should be used.
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/.cvsignore:
+ * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
+ (gio_testsuite), (main):
+ Add unit test for giostreamsrc and giostreamsink.
+
+2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (plugin_init):
+ Remove nowadays unnecessary workaround for a crash.
+ * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
+ (gst_gio_sink_start), (gst_gio_sink_stop),
+ (gst_gio_sink_unlock_stop):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
+ (gst_gio_src_stop), (gst_gio_src_unlock_stop):
+ * ext/gio/gstgiosrc.h:
+ Make the finalize function safer, clean up everything that could stay
+ around.
+ Reset the cancellable instead of creating a new one after cancelling
+ some operation.
+ Don't store the GFile in the element, it's only necessary for creating
+ the streams.
+
+2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
+ Original commit message from CVS:
+ Patch by: Sebastien Moutte <sebastien moutte net>
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
+ (gst_rtcp_unix_to_ntp):
+ * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
+ Fix some C99-isms and and a missing function that some versions of
+ MSVC don't like too much (#494346).
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstaudio.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgsttag.dsp:
+ Update vs6 projects files (#494346).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ win32/common/: More missing symbols to export (fixes #493986).
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstcdda.def:
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstpbutils.def:
+ * win32/common/libgstrtp.def:
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgsttag.def:
+ * win32/common/libgstvideo.def:
+ More missing symbols to export (fixes #493986).
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_link),
- (gst_deinterlace_chain):
- Rowstride fixes. Fixes #161039.
- * gst/videocrop/gstvideocrop.c: (gst_video_crop_init),
- (gst_video_crop_get_property), (gst_video_crop_add_to_struct_val),
- (gst_video_crop_getcaps), (gst_video_crop_link),
- (gst_video_crop_i420), (gst_video_crop_chain),
- (gst_video_crop_change_state):
- Rework of negotiation. Actually works now. Fixes #158650.
+2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/fft/gstfftf32.c:
+ * gst-libs/gst/fft/gstfftf32.h:
+ * gst-libs/gst/fft/gstfftf64.c:
+ * gst-libs/gst/fft/gstfftf64.h:
+ * gst-libs/gst/fft/gstffts16.c:
+ * gst-libs/gst/fft/gstffts16.h:
+ * gst-libs/gst/fft/gstffts32.c:
+ * gst-libs/gst/fft/gstffts32.h:
+ * tests/check/libs/fft.c: (GST_START_TEST):
+ Remove the magnitude and phase calculation functions as these have
+ very special use cases and can't even be used for the spectrum
+ element. Also adjust the docs to mention some properties of the used
+ FFT implemention, i.e. how the values are scaled. Fixes #492098.
+
+2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
+ (finish_source):
+ Avoid crash when there are external subtitles (fixes #491722).
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_sint):
- That was very stupid.
+ ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
+ 'Could not open resource for writing' is not an acceptable
+ error message when we can't open the audio device (see #492334),
+ even less so when we're trying to open it to record something.
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix possible crasher.
+ win32/common/libgstrtp.def: Add some more missing symbols (#492813).
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * win32/common/libgstrtp.def:
+ Add some more missing symbols (#492813).
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_uint),
- (gst_matroska_ebmlnum_sint), (gst_matroska_demux_parse_blockgroup):
- Lace sizes can be zero.
+ tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
+ Original commit message from CVS:
+ Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * tests/check/elements/audioconvert.c: (verify_convert):
+ Add check to make sure that the out caps have a channel layout
+ set on them where they should have one.
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
- * ext/musepack/gstmusepackdec.cpp:
- Fetch error return values. Fixes #161624.
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
- Really EOS.
+ gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
+ Original commit message from CVS:
+ Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+ * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
+ * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
+ Include our own _stdint.h instead of sys/types.h, makes MingW happy
+ (#492306).
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
+ Use _pipe directly, GLib doesn't have a pipe() macro any longer
+ (it disappeared in GLib 2.14.0) (#492306).
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Fix includes and LIBS for win32/Mingw (#492306).
+ * tests/examples/dynamic/addstream.c (pause_play_stream):
+ Use more portable g_usleep() instead of sleep() (#492306).
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
- Work for truncated (unfinished download etc.) files. Fixes #160514.
+ gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
+ (gst_ring_buffer_parse_caps):
+ Return NULL instead of an enum that happens to be 0, fixes warning
+ on MSVC (#492114).
+ * gst-libs/gst/audio/gstringbuffer.h:
+ No trailing commas in enum list (for gcc-2.9x).
+ * gst/videotestsrc/videotestsrc.c: (random_char):
+ Make information loss explicit instead of implicitly truncating to
+ eight bits via the return value. Fixes runtime error on MSVC when
+ using the debug CRT (#492114).
+ * win32/common/config.h.in:
+ Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstrtp.def:
+ Export a few more symbols (#492114).
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Fix for integer overflow. Makes #156001 not crash. Probably masks
- the real bug.
+ gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/audio.h:
+ Readd the deprecation guards, but preserve compilability.
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ac3parse/gstac3parse.c: (plugin_init):
- Parsers never have ranks. Fixes #159651.
+ gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
+ (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
+ Preserve channel layout when fixating the number of channels in the
+ output caps, or make sure there's a suitable channel position layout
+ set on the caps if required. Fixes #430677.
-2004-12-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstdecodebin.c: (compare_ranks):
- make sure the facotries are ordered the same every time even if they
- have the same rank by using the name
- * gst/playback/gstdecodebin.c: (find_compatibles):
- make sure we don't add factories to the list twice
+ tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
+ Original commit message from CVS:
+ * tests/check/elements/decodebin.c: (test_text_plain_streams):
+ Make sure the pipeline really operates in push mode as it should
+ in this case.
-2004-12-16 David Schleef <ds@schleef.org>
+2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: look for musepack headers as musepack/*.h
- (fixes #159847)
- * ext/musepack/gstmusepackdec.h: use <musepack/*.h>
- * ext/musepack/gstmusepackreader.h: same
+ gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h:
+ Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
+ compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
+ (ie. normal cvs builds) will fail.
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_auds_with_data):
- Read extradata correctly (fixes #155879).
+ tell gtk-doc about the deprecation guard. Apply more doc fixes.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/interfaces/mixer.c:
+ tell gtk-doc about the deprecation guard. Apply more doc fixes.
-2004-12-16 David Schleef <ds@schleef.org>
+2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioscale/gstaudioscale.c: allow passthru of >2 channel
- audio. does _not_ attempt or allow conversion unless channels
- is 1 or 2.
+ tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
+ Original commit message from CVS:
+ * tests/check/libs/audio.c: (init_value_to_channel_layout),
+ (test_channel_layout_value_intersect), (audio_suite):
+ Add simple unit test to make sure GstValue intersection
+ of channel layouts works the way I think it does.
-2004-12-16 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * tools/gst-launch-ext-m.m: fix mpeg and vob pipelines
+ Fix the docs according to what gtk-doc complained about.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Fix the docs according to what gtk-doc complained about.
-2004-12-16 David Schleef <ds@schleef.org>
+2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioscale/gstaudioscale.c: the resample library only
- handles 1 or 2 channels. Change caps to compensate.
+ tests/icles/stress-playbin.c: Fix the build.
+ Original commit message from CVS:
+ * tests/icles/stress-playbin.c:
+ Fix the build.
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/matroska/matroska-demux.c: (aac_rate_idx), (aac_profile_idx),
- (gst_matroska_demux_audio_caps):
- Some MPEG-AAC hacks, because else it doesn't work...
+ gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad):
+ Post nice/more useful error message if we don't have a decoder for
+ the primary type.
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
+ Be a bit more useful, unblock the pads after we fired the no-more-pads
+ signal so that we can use the signal to inspect and connect all pads
+ without having to keep extra state outside of decodebin.
+
+2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c:
+ (gst_uri_decode_bin_autoplug_continue),
+ (gst_uri_decode_bin_class_init), (no_more_pads_full):
+ Implement default signal handler so that we return TRUE when nothing is
+ connected.
+
+2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c:
+ (gst_riff_wavext_add_channel_layout),
+ (gst_riff_wave_add_default_channel_layout),
+ (gst_riff_wavext_get_default_channel_mask),
+ (gst_riff_create_audio_caps):
+ Use the ALSA channel layout as default for wav files without channel
+ layout information. This fixes playback of chan-id.wav on 5.1 systems
+ for example. Also refactor the channel layout setting a bit and add
+ more default channel orders. Fixes #489010.
+
+2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add h264.
+ * ChangeLog:
+ Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
+ Original commit message from CVS:
+ (gst_riff_wavext_add_channel_layout),
+ (gst_riff_wave_add_default_channel_layout),
+ (gst_riff_wavext_get_default_channel_mask),
+ (gst_riff_create_audio_caps):
+ Use the ALSA channel layout as default for wav files without channel
+ layout information. This fixes playback of chan-id.wav on 5.1 systems
+ for example. Also refactor the channel layout setting a bit and add
+ more default channel orders. Fixes #489010.
+
+2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
+ Original commit message from CVS:
+ * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
+ GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
+ -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
+ instead.
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
- * gst-libs/gst/audio/Makefile.am:
- Try to fix buildbot.
+ * gst-plugins-base.spec.in:
+ update spec file
+ Original commit message from CVS:
+ update spec file
+
+2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
+ (gst_decode_bin_set_subs_encoding),
+ (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
+ (gst_decode_bin_get_property), (analyze_new_pad):
+ Move subtitle encoding property to decodebin2 so that it can set the
+ property value on all elements that it autoplugs and that require it.
+ Make caps refcounting more consistent in get/set.
+ * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
+ (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
+ (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
+ (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
+ (proxy_autoplug_continue_signal),
+ (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
+ (make_decoder):
+ Proxy properties and relevant signals from the internal decodebin.
+ Make properties MT safe.
+
+2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
+ * gst-libs/gst/tag/tags.c:
+ Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
+ GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
+ * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
+ Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
+ * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
+ (gst_tag_to_vorbis_comments):
+ Map new SORTNAME tags (these tags aren't even semi-official, so I'm
+ just mapping everything I found in the wild) (#414539).
-2004-12-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
+ Original commit message from CVS:
+ Inspired by patch of: René Stadler <mail at renestadler dot de>
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_autoplug_continue),
+ (gst_decode_bin_autoplug_factories),
+ (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
+ (find_compatibles):
+ * gst/playback/gstplay-marshal.list:
+ Remove the autoplug-sort signal and replace it with a binding friendly
+ autoplug-select signal.
+ Add an autoplug-factories signal that can be used to generate a list of
+ factories to try to autoplug.
+ Add the GstPad to the autoplugging signal args as it might be needed to
+ make a good factory selection.
+ Fix up the marshallers for this. Fixes #407282.
+
+2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't abort with an assertion if we receive a seek event with
+ a start type of NONE (see launchpad bug #155878).
- * gst/tcp/gstmultifdsink.c:
- Clean up and uniformize debugging.
+2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
+ (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
+ (gst_ximagesink_change_state), (gst_ximagesink_reset):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
+ (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
+ (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
+ Make sure that before we clean up the X resources, we shutdown and join
+ the event thread.
+ Also make sure the event thread does not shut down immediatly after
+ startup because the running variable is not yet correctly set.
+ Fixes #378770.
+
+2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (new_pad), (type_found):
+ Make the window for a race in typefind and shutting down smaller until
+ we figure out the right locking here. Avoids #485753 usually.
+ * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
+ Remove unneeded lock causing a race in typefind and shutting down.
+ Fixes #485753.
+ * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
+ Also remove sinks when going to NULL because we might not complete the
+ state change to PAUSED, causing the PAUSED->READY state change not to
+ happen.
+
+2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
+ Also explicitly release the ringbuffer when going to NULL because it
+ is required in the setcaps function, before the state change to PAUSED
+ completes.
+
+2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/icles/: Does what it says on the tin.
+ Original commit message from CVS:
+ * tests/icles/.cvsignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/stress-playbin.c:
+ Does what it says on the tin.
-2004-12-16 Edward Hervey <bilboed@bilboed.com>
+2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
+ Fix queue negotiation. See #486758.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
- (gst_dvd_demux_reset), (gst_dvd_demux_change_state):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_reset),
- (gst_mpeg_demux_change_state):
- Reset on ready. Fixes 160276.
+2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+ Actual code change to go along with:
+ Original commit message from CVS:
+ Actual code change to go along with:
+ 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
+ (gst_xvimagesink_xwindow_new),
+ (gst_xvimagesink_update_colorbalance),
+ (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
+ Fix handling of some of the X atoms. If the last parameter is True,
+ XInternAtom won't create the atom if it doesn't exist, and therefore
+ might return None. This causes X errors on Xv implementations that
+ don't provide the colour balance attributes.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_pad_link):
- Fix memleak (#154815).
+ * ChangeLog:
+ Remove stray character from the changelog.
+ Original commit message from CVS:
+ Remove stray character from the changelog.
-2004-12-16 James Bowes <bowes@cs.dal.ca>
+2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
- * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init),
- (gst_musicbrainz_init), (gst_musicbrainz_chain),
- (gst_musicbrainz_set_property), (gst_musicbrainz_get_property):
- * ext/musicbrainz/gsttrm.h:
- Add support for using a proxy server when getting a trm id from
- the MusicBrainz database (#149613).
+2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-16 Christophe Fergeau <teuf@gnome.org>
+ Extract vorbis comment LICENSE tags correctly.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * tests/check/libs/tag.c:
+ Extract vorbis comment LICENSE tags correctly.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
- * gst/playback/gstdecodebin.c: (new_pad), (close_link):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Fix memleaks (#157233).
+ Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
+ Original commit message from CVS:
+ Patch by: Jason Kivlighn <jkivlighn gmail com>
+ * gst-libs/gst/tag/gstid3tag.c:
+ * tests/check/libs/tag.c:
+ Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't error out when a buggy downstream element doesn't
+ handle the newsegment event we send properly (especially
+ not without posting a meaningful error message on the
+ bus). See bug #471370 and launchpad bug #136264.
- * gst-libs/gst/resample/resample.c: (gst_resample_close):
- * gst-libs/gst/resample/resample.h:
- * gst/audioscale/gstaudioscale.c:
- Fix memleak (#159215).
+2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain):
+ Use new basesink method to make our EOS drain interruptable.
+
+2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ Fix silly search-replace oversight.
+
+2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
+ Original commit message from CVS:
+ Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+ * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
+ (gst_basertppayload_set_outcaps):
+ Fix caps memleak. Fixes #484989.
+
+2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
+ Fix debug output.
+
+2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Also handle the case where there is no clock set on the audio source,
+ like in the unit tests.
+
+2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
+ to avoid compiler warnings
+
+2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (type_found),
+ (gst_decode_bin_change_state):
+ * gst/playback/gstdecodebin2.c: (type_found),
+ (gst_decode_bin_change_state):
+ Don't disconnect the have_type signal because we never reconnect it
+ later on. Instead keep a variable to see if we already detected a type.
+
+2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
+ (type_found):
+ Unlink the signal handler when we found the type, we're not going to do
+ anything sensible with more type_found signals anyway.
+
+2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Use GIO function to get a list of supported URI schemes instead of
+ hard coding something.
+
+2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't leak caps.
-2004-12-16 Toni Willberg <toniw@iki.fi>
+2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ * gst-libs/gst/tag/gsttagdemux.h:
+ API: add GstTagDemux base class for simple tag demuxers.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add GstTagDemux to docs.
- * sys/oss/gstosselement.c: (gst_osselement_probe_caps):
- * sys/oss/oss_probe.c: (main):
- Check for mono/stereo support (similar to samplerate probing),
- fixes #159433. Also add missing copyright header to oss_probe.c.
+2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-12-15 David Schleef <ds@schleef.org>
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_get_payload_subbuffer):
+ Fix bug introduced with last commit which inverted the logic and
+ caused all buffers to be dropped. Fixes #483620.
+ Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
- * configure.ac: add audioresample and cairo plugins. Remove
- HAVE_MMX stuff, because it's not used.
- * ext/Makefile.am: same
- * ext/audioresample/Makefile.am: You are not ready for an
- audio resampling element based on audioresample.
- * ext/audioresample/gstaudioresample.c:
- * ext/audioresample/gstaudioresample.h:
- * ext/cairo/Makefile.am: You are not ready for overlay elements
- based on cairo. Don't look too closely, these elements kinda
- suck right now.
- * ext/cairo/gstcairo.c: new
- * ext/cairo/gsttextoverlay.c: new
- * ext/cairo/gsttextoverlay.h: new
- * ext/cairo/gsttimeoverlay.c: new
- * ext/cairo/gsttimeoverlay.h: new
- * gst-libs/gst/media-info/media-info-priv.h: fix compile
- problem with compilers that don't support variadic macros.
+2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-15 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+ gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ Replace g_return_if_val (as it could be disabled), with regular return
+ and warning.
- Reviewed by: David Schleef <ds@schleef.org>
+2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/sunaudio/gstsunaudio.c: (plugin_init): Apply patch from
- Bala, registering sunaudiosrc (oops!), and cleaning up code a
- bit. Also ran indent-gst.
- * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_init),
- (gst_sunaudiosrc_change_state), (gst_sunaudiosrc_get),
- (gst_sunaudiosrc_setparams):
+ tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
+ Original commit message from CVS:
+ * tests/check/pipelines/simple-launch-lines.c:
+ Print message name and not just number.
-2004-12-14 David Schleef <ds@schleef.org>
+2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/festival/gstfestival.c: (gst_festival_chain): Set the
- output rate to 16000. Should fix #160235.
+ gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_async_play):
+ When slaved to the clock, don't try to align a sample with the previous
+ one when going to PLAYING again.
-2004-12-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Add typefinding for mpeg2 pes streams
+ tests/examples/snapshot/snapshot.c: Fix the build.
+ Original commit message from CVS:
+ * tests/examples/snapshot/snapshot.c:
+ Fix the build.
-2004-12-13 David Schleef <ds@schleef.org>
+2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgiosink.c: Update to API changes in GIO.
+ Original commit message from CVS:
+ * ext/gio/gstgiosink.c: (gst_gio_sink_start):
+ Update to API changes in GIO.
+
+2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
+ Original commit message from CVS:
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ Add RFC 3556 bandwidth modifiers.
+
+2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Update documentation.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ Update documentation.
+
+2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/: Added new file and header to deal with payload info.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
+ (gst_rtp_payload_info_for_name):
+ * gst-libs/gst/rtp/gstrtppayloads.h:
+ Added new file and header to deal with payload info.
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
+ (gst_rtp_buffer_default_clock_rate):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Payload specific stuff is move to new headers.
+ Implement _default_clock rate using the new payload function.
+ * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
+ (gst_sdp_parse_line):
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ Add some more comments.
+
+2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
+ (sdp_check_header), (sdp_type_find), (plugin_init):
+ Add typefind function for application/sdp.
+ Remove some old dirac typefind code that was ifdeffed out.
+
+2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/common/libgstaudio.def: Add new exported functions.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add new exported functions.
+ * win32/vs6/grammar.dsp:
+ Add autogeneration and copy of some autegenerated files from win32/common
+ for rtsp library.
+ * win32/vs6/libgstaudioconvert.dsp:
+ Add gstaudioquantize.c to the build.
+ * win32/vs6/libgstinterfaces.dsp:
+ Add videoorientation.c to the build.
+ * win32/vs6/libgstriff.dsp:
+ Add libgsttag to the link libraries list.
+ * win32/vs6/libgstvolume.dsp:
+ Add liboil to the link.
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstrtsp.dsp:
+ * win32/common/libgstrtsp.def:
+ Add files to build libgstrtsp library.
+
+2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
+ Original commit message from CVS:
+ * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
+ (gst_gio_sink_set_property), (gst_gio_sink_render):
+ * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
+ (gst_gio_src_set_property):
+ Some minor cleanup and allow setting the location only when the
+ element is not playing or paused.
+
+2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
+ Original commit message from CVS:
+ * tests/examples/snapshot/snapshot.c: (main):
+ Print error when pipeline failed to construct.
+
+2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Add mappings for the new GST_TAG_COMPOSER for vorbis comments
+ and ID3v2 tags.
- * configure.ac: Applied patch from bug #143659, making default
- sources and sinks OS-dependent (for Solaris), and added code
- for OS/X.
- * gconf/gstreamer.schemas.in: use OS-dependent sinks in gconf.
+2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
+ Original commit message from CVS:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Don't include config.h in an installed public header, this
+ might break compilation of applications that don't have such
+ a header and doesn't necessarily do what it's supposed to do
+ anyway (ie. check for the lrint/lrintf defines) (#442065).
+ Add docs for the various macros and document how this header
+ has to be used (link against libm, etc.); add a few FIXMEs;
+ include math.h for non-c99 code path. Based on patch by
+ Jan Schmidt.
- * gst-libs/gst/riff/riff-media.c:
- forgot to add h2.64 to avidemux template caps
+2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
+ Original commit message from CVS:
+ * configure.ac:
+ Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
+ of duplicating these macros in configure.ac.
- * gst/wavenc/riff.h:
- * gst-libs/gst/riff/riff-media.c:
- * gst-libs/gst/riff/riff-ids.h:
- * gst/avi/gstavimux.c
- add 4CC code for VideoSoft h264 in AVI (VSSH)
- fixes bug #160655
- remove s323 from riff, it's quicktime specific :(
+2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ po/: Updated translations to 0.10.14
+ Original commit message from CVS:
+ * po/hu.po:
+ * po/sv.po:
+ * po/uk.po:
+ Updated translations to 0.10.14
- * gst/asfdemux/README
- * gst/wavenc/riff.h
- * gst-libs/gst/riff/riff-ids.h
- * gst-libs/gst/riff/riff-media.c
- * gst/qtdemux/qtdemux.c:
- add new 4CC codes for h263 related codecs
- fixes partially bug #155163
+2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-12 Christian Fredrik Kalager Schaller <christian at fluendo dot com>
+ * po/LINGUAS:
+ add languages
+ Original commit message from CVS:
+ add languages
- * configure.ac: Update polyaudio requirement to 0.7
- * ext/polyp/polypsink.c: (create_stream): add patch from iain (158258)
+2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ po/pl.po: Added Polish translation.
+ Original commit message from CVS:
+ translated by: Jakub Bogusz <qboosh@pld-linux.org>
+ * po/pl.po:
+ Added Polish translation.
- * gst/interleave/deinterleave.c:
- fix my name's spelling! :)
+2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ po/fi.po: Added Finnish translation.
+ Original commit message from CVS:
+ translated by: Ilkka Tuohela <hile@iki.fi>
+ * po/fi.po:
+ Added Finnish translation.
- * AUTHORS ChangeLog
- * gst/auparse/gstauparse.c
- * gst/interleave/deinterleave.c
- * gst/law/:
- alaw-decode.c alaw-encode.c
- mulaw-decode.c mulaw-encode.c
- * gst/oneton/gstoneton.c
- * sys/osxaudio/:
- gstosxaudioelement.c gstosxaudiosink.c gstosxaudiosrc.c
- * sys/osxvideo/:
- cocoawindow.h cocoawindow.m
- osxvideosink.h osxvideosink.m
+2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- put the same mail address for Zaheer Abbas Merali everywhere
+ po/es.po: Added Spanish translation.
+ Original commit message from CVS:
+ translated by: Jorge González González <aloriel@gmail.com>
+ * po/es.po:
+ Added Spanish translation.
-2004-12-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
- Align by packetsize, and assert that we a packet available before
- playing. The first makes webstreams work (they often include
- trailing padding data in a packet), the second allows pausing a
- ASF stream in totem without getting demux errors afterwards.
+ po/da.po: Added Danish translation.
+ Original commit message from CVS:
+ translated by: Mogens Jaeger <mogens@jaeger.tf>
+ * po/da.po:
+ Added Danish translation.
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c: (get_relative):
- Check for non-NULL before accessing member (end-of-chain).
+ po/zh_CN.po: Added Chinese (simplified) translation.
+ Original commit message from CVS:
+ translated by: Funda Wang <fundawang@linux.net.cn>
+ * po/zh_CN.po:
+ Added Chinese (simplified) translation.
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_set_property), (cdparanoia_get_property):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init),
- (dvdnavsrc_set_property), (dvdnavsrc_get_property):
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
- (dvdreadsrc_init), (dvdreadsrc_set_property),
- (dvdreadsrc_get_property):
- * sys/vcd/vcdsrc.c: (gst_vcdsrc_class_init),
- (gst_vcdsrc_set_property), (gst_vcdsrc_get_property):
- Synchronize property names where not yet the case. Devices are
- now device=X, other versions are deprecated (but still exist).
- Also use g_free() unconditionally.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (setup_source), (gst_play_base_bin_get_property):
- Expose source.
+ po/bg.po: Added Bulgarian translation.
+ Original commit message from CVS:
+ translated by: Alexander Shopov <ash@contact.bg>
+ * po/bg.po:
+ Added Bulgarian translation.
-2004-12-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-bad-plugins.hierarchy:
+ Update hierarchy.
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.h:
+ Mark private fields of the instance structs private.
+
+2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * docs/plugins/gst-plugins-bad-plugins.args:
+ * docs/plugins/gst-plugins-bad-plugins.signals:
+ * docs/plugins/inspect/plugin-bz2.xml:
+ * docs/plugins/inspect/plugin-cdxaparse.xml:
+ * docs/plugins/inspect/plugin-dfbvideosink.xml:
+ * docs/plugins/inspect/plugin-dtsdec.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-faac.xml:
+ * docs/plugins/inspect/plugin-faad.xml:
+ * docs/plugins/inspect/plugin-filter.xml:
+ * docs/plugins/inspect/plugin-freeze.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gsm.xml:
+ * docs/plugins/inspect/plugin-gstrtpmanager.xml:
+ * docs/plugins/inspect/plugin-h264parse.xml:
+ * docs/plugins/inspect/plugin-modplug.xml:
+ * docs/plugins/inspect/plugin-mpeg2enc.xml:
+ * docs/plugins/inspect/plugin-musepack.xml:
+ * docs/plugins/inspect/plugin-musicbrainz.xml:
+ * docs/plugins/inspect/plugin-nsfdec.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-soundtouch.xml:
+ * docs/plugins/inspect/plugin-spcdec.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speed.xml:
+ * docs/plugins/inspect/plugin-tta.xml:
+ * docs/plugins/inspect/plugin-videosignal.xml:
+ * docs/plugins/inspect/plugin-xingheader.xml:
+ * docs/plugins/inspect/plugin-xvid.xml:
+ Add the GIO plugin to the docs and do a make update
+ while doing that.
+ * ext/gio/gstgiosrc.c: (gst_gio_src_start):
+ Fix a small memleak.
+
+2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
+
+ Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler dot de>
+ * configure.ac:
+ * ext/Makefile.am:
+ * ext/gio/Makefile.am:
+ * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
+ (gst_gio_get_supported_protocols),
+ (gst_gio_uri_handler_get_type_sink),
+ (gst_gio_uri_handler_get_type_src),
+ (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
+ (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
+ (gst_gio_uri_handler_do_init), (plugin_init):
+ * ext/gio/gstgio.h:
+ * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
+ (gst_gio_sink_class_init), (gst_gio_sink_init),
+ (gst_gio_sink_finalize), (gst_gio_sink_set_property),
+ (gst_gio_sink_get_property), (gst_gio_sink_start),
+ (gst_gio_sink_stop), (gst_gio_sink_unlock),
+ (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
+ (gst_gio_sink_render), (gst_gio_sink_query):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
+ (gst_gio_src_class_init), (gst_gio_src_init),
+ (gst_gio_src_finalize), (gst_gio_src_set_property),
+ (gst_gio_src_get_property), (gst_gio_src_start),
+ (gst_gio_src_stop), (gst_gio_src_get_size),
+ (gst_gio_src_is_seekable), (gst_gio_src_unlock),
+ (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
+ (gst_gio_src_create):
+ * ext/gio/gstgiosrc.h:
+ Add a GIO/GVFS plugin with source and sink elements. This will
+ only be enabled when --enable-experimental is given to configure
+ for now as the GIO API is not stable yet. Fixes #476916.
+
+2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_push_one):
+ Fix compilation wrt printf arguments.
+
+2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
+ Original commit message from CVS:
+ * examples/app/appsrc_ex.c: (main):
+ Fix compilation after changing the name of a method.
+
+2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add simple snapshot example program using appsink.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/snapshot/.cvsignore:
+ * tests/examples/snapshot/Makefile.am:
+ * tests/examples/snapshot/snapshot.c: (main):
+ Add simple snapshot example program using appsink.
+
+2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
+ (gst_app_sink_class_init), (gst_app_sink_init),
+ (gst_app_sink_dispose), (gst_app_sink_finalize),
+ (gst_app_sink_set_property), (gst_app_sink_get_property),
+ (gst_app_sink_flush_unlocked), (gst_app_sink_start),
+ (gst_app_sink_event), (gst_app_sink_getcaps),
+ (gst_app_sink_set_caps), (gst_app_sink_get_caps),
+ (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
+ (gst_app_sink_pull_buffer):
+ * gst-libs/gst/app/gstappsink.h:
+ Add properties, signals and actions to access the element even without
+ linking to the library.
+ Fix some method names and signatures.
+
+2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/generic/states.c: Improved state change unit test.
+ Original commit message from CVS:
+ * tests/check/generic/states.c:
+ Improved state change unit test.
- * configure.ac: move GCONF macro outside conditional for the am
- conditional. Fixes #160439
+2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-08 David Schleef <ds@schleef.org>
+ Ignore registries in any format.
+ Original commit message from CVS:
+ * docs/plugins/.cvsignore:
+ * tests/check/.cvsignore:
+ Ignore registries in any format.
- * tools/gst-visualise-m.m: Switch to elements that currently
- exist.
+2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Only copy timestamp on outgoing packets if the depayloader did not set
+ one.
+ Also copy duration on outgoing packets.
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- We love wrong commas.
+2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
+ (gst_basertppayload_set_outcaps):
+ Fix compilation because of missing %d in printf.
+ When fixating caps, fixate what we can and throw away all remaining
+ unfixed caps, subclasses should do something smart if they need to.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_src_query):
- Don't set DEFAULT, unsupported - makes length display incorrectly
- in some cases.
+2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Improve debug logs a bit and be more verbose if things go wrong.
- * gst/monoscope/README: remove blurb about files being GPL
- * gst/monoscope/gstmonoscope.c: Change license field to LGPL
- * gst/monoscope/monoscope.c: Change license to BSD with explanation
- monoscope is now effectively LGPL licensed
+2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ Fix a bunch of compile warnings shown with Forte.
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_set_property):
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
+ (gst_rtcp_unix_to_ntp):
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
+ * gst/playback/gstqueue2.c:
+ * tests/examples/seek/seek.c: (set_scale):
+ Fix a bunch of compile warnings shown with Forte.
+ * gst/audiorate/gstaudiorate.c:
+ Always pull in config.h before including any system headers.
+
+2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Also fix #476514 for queue2.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (update_buffering),
+ (gst_queue_locked_flush), (gst_queue_locked_enqueue),
+ (gst_queue_handle_sink_event), (gst_queue_chain),
+ (gst_queue_push_one), (gst_queue_sink_activate_push),
+ (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
+ Also fix #476514 for queue2.
+
+2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_handle_sink_event),
+ (gst_base_rtp_depayload_push_full),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state):
+ Remove code to deal with RTP to GST time conversion, we now just copy
+ the GST timestamp we receive to the outgoing buffers.
+ Handle segment and flushes correctly.
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ When we have no valid input timestamp, use the previous rtp timestamp on
+ the outgoing RTP packet instead of the RTP base time.
+
+2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
+
+ ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c:
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ Change alsa alloca's to malloc to fix warnings on gcc-4.2.
- * gst/monoscope/README: Update information to be more correct
- * gst/monoscope/convolve.c: Relicense to LGPL
- * gst/monoscope/convolve.h: Relicense to LGPL
+2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
+ Add some debug info when negotiating caps.
+
+2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
+ A buffer with an empty payload is also a valid buffer.
+
+2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
+ (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
+ (gst_basertppayload_change_state):
+ Make sure we start our RTP timestamp from the random base RTP
+ timestamp even if the buffer timestamp starts from some random value.
+
+2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/dynamic/.cvsignore:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/dynamic/addstream.c: (create_stream),
+ (pause_play_stream), (message_received), (eos_message_received),
+ (perform_step), (main):
+ Add simple exmple app to demonstrate starting and pausing live and
+ non-live bins in a PLAYING pipeline.
+
+2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
+ Original commit message from CVS:
+ 2007-09-14 Julien MOUTTE <julien@moutte.net>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
+ typefind for QCP files (RFC #3625)
+
+2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_init):
+ Disable pull mode scheduling, we're not ready for it yet and it subtly
+ breaks a lot of things.
+
+2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
+ Original commit message from CVS:
+ * tests/check/elements/libvisual.c:
+ Test all libvisual plugins, not just the first one; this reproduces
+ bug #450336 quite easily. Looks like a problem with the 'jess'
+ visualisation.
-2004-12-06 Arwed v. Merkatz <v.merkatz@gmx.net>
+2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- set BUFFER_DURATION to correct values (mpeg1 audio frame length is fixed)
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link):
- set default_duration for mpeg1 audio
+ tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/libvisual.c:
+ Add basic libvisual test case in an attempt to reproduce bug #450336.
+ Doesn't reproduce that bug, but some other crasher instead (invalid
+ free), at least with make elements/libvisual.forever and the bumscope
+ plugin on x86-64/gutsy. Leaving test disabled for now.
-2004-12-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ gst/: Printf format fixes (#476128).
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/app/gstappsink.c:
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvparse.c:
+ * gst/interleave/deinterleave.c:
+ * gst/switch/gstswitch.c:
+ Printf format fixes (#476128).
+
+2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
+ (read_body), (gst_rtsp_connection_receive):
+ Make sure we can not cancel in the middle of receiving a message.
+ Fixes #475731.
+
+2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
+
+ gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
+ Original commit message from CVS:
+ Patch by: Josep Torra Valles <josep@fluendo.com>
+ * gst/playback/gstplaybasebin.c:
+ Increase upper limit for audio queue a bit; fixes preroll problem
+ with playbin and decodebin2 when playing a quicktime trailer with
+ multichannel audio via http (#464666).
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_close_audio):
- * ext/alsa/gstalsa.h:
- refactor big chunks of the core caps negotiation code to make it
- a lot faster, because people claim it's really slow
- (actually, just cache the getcaps when the device is opened)
-
-2004-12-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_init),
- (gst_a52dec_handle_event), (gst_a52dec_update_streaminfo),
- (gst_a52dec_handle_frame), (gst_a52dec_chain),
- (gst_a52dec_change_state), (plugin_init):
- * ext/a52dec/gsta52dec.h:
- Do something useful with timestamps. Make chain-based (since
- there's really no reason to be loopbased).
- * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
- Update current_byte/frame correctly.
-
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/apetag/apedemux.c: (gst_ape_demux_parse_tags),
- (gst_ape_demux_stream_init):
- Forward tags, too.
-
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_init):
- Let's make sure we're done typefinding when detecting tags.
-
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/ebml-read.c: (gst_ebml_read_class_init),
- (gst_ebml_read_init), (gst_ebml_read_use_event),
- (gst_ebml_read_element_id), (gst_ebml_peek_id),
- (gst_ebml_read_seek), (gst_ebml_read_skip),
- (gst_ebml_read_reserve), (gst_ebml_read_buffer),
- (gst_ebml_read_master):
- * gst/matroska/ebml-read.h:
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_contents),
- (gst_matroska_demux_loop_stream), (gst_matroska_demux_audio_caps):
- Disgustingly evil hack for working around INTERRUPT events and
- their extremely annoying habit of being a pain in the ass. We
- simply peek a cluster before reading any of it.
-
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/musepack/gstmusepackdec.cpp:
- There's also floating point libmusepacks.
-
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
- (gst_faad_chanpos_to_gst), (gst_faad_chain):
- Set DURATION even if source buffer didn't. Also use increasing
- timestamps.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Block_align can have larger values than 8192.
+2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init), (gst_base_audio_src_init),
+ (gst_base_audio_src_provide_clock),
+ (gst_base_audio_src_set_property),
+ (gst_base_audio_src_get_property), (gst_base_audio_src_create):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ Allow othe clocks than the internal clock to be used for the pipeline.
+ Add property to disable clock provide.
+ API: GstBaseAudioSrc::provide-clock
+
+2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Don't leak request pads. Fixes #475395.
-2004-12-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
- * gst/law/alaw-decode.c: (alawdec_getcaps), (alawdec_link):
- * gst/law/alaw-encode.c: (alawenc_getcaps), (alawenc_link):
- * gst/law/mulaw-decode.c: (mulawdec_getcaps), (mulawdec_link):
- * gst/law/mulaw-encode.c: (mulawenc_getcaps), (mulawenc_link):
- Fix caps.
+ sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler dot de>
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximage_buffer_class_init):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_class_init):
+ Correctly chain up finalize with the parent class to prevent
+ memory leaks. Fixes #474880.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix logic bug.
+ Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func):
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ Revert the latest change: floating point samples are allowed to
+ have any value, not only values in the range [-1,1]. Thanks to Andy
+ Wingo for noticing.
+ Also fix processing of int32 samples with volumes > 4 by making the
+ unity value smaller which prevents overflows.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Yay, another one.
+ Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * tests/check/libs/rtp.c:
+ Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
- * ext/esd/esdsink.c: (gst_esdsink_chain):
- Make error actually say something useful (fixes #156798).
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add Intel Video 5.0 fourcc (IV50).
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
+ Original commit message from CVS:
+ Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ Fix up GstRTPHeader helper struct so that compilers will not under
+ any circumstances add padding in between our fields, as currently
+ happens with MSVC on win32, because that would lead to us sending
+ out RTP payloads with broken RTP headers (#471194).
+ Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/rtp.c:
+ Add some simple unit tests for GstRTPBuffer. Some are disabled
+ because the code tested still needs fixing (set_csrc() does not work).
-2004-12-01 Christophe Fergeau <teuf@gnome.org>
+2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
- * ext/mad/gstmad.c: (mpg123_parse_xing_header): fix xing header
- detection on mono and stereo mp3 files.
+ * gst-plugins-base.spec.in:
+ update spec file to include latest RTSP libraries and headers and more
+ Original commit message from CVS:
+ update spec file to include latest RTSP libraries and headers and more
+
+2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ win32/: Add rtsp enumtypes (#474384) and update others.
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/gstrtsp-enumtypes.h:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ Add rtsp enumtypes (#474384) and update others.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Don't crash on EMPTY caps (e.g. when the demuxer didn't recognize
- the contained stream).
+ configure.ac: Fix configure check for HAVE_LIBXML_HTML.
+ Original commit message from CVS:
+ * configure.ac:
+ Fix configure check for HAVE_LIBXML_HTML.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_srcconnect), (gst_faad_chain):
- Oops, remove debug.
+ tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
+ Original commit message from CVS:
+ * tests/check/libs/.cvsignore:
+ Ignore more, in case the build bots work again one day.
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add libgstfft, a FFT library based on Kiss FFT which is
+ Original commit message from CVS:
+ Reviewed by: Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
+ * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
+ * gst-libs/gst/fft/gstfft.h:
+ * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
+ (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
+ (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
+ * gst-libs/gst/fft/gstfftf32.h:
+ * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
+ (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
+ (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
+ * gst-libs/gst/fft/gstfftf64.h:
+ * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
+ (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
+ (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
+ * gst-libs/gst/fft/gstffts16.h:
+ * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
+ (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
+ (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
+ * gst-libs/gst/fft/gstffts32.h:
+ * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
+ (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
+ (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
+ (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
+ * gst-libs/gst/fft/kiss_fft_f32.h:
+ * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
+ (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
+ (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
+ (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
+ * gst-libs/gst/fft/kiss_fft_f64.h:
+ * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
+ (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
+ (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
+ (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
+ * gst-libs/gst/fft/kiss_fft_s16.h:
+ * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
+ (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
+ (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
+ (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
+ * gst-libs/gst/fft/kiss_fft_s32.h:
+ * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
+ (kiss_fftr_f32), (kiss_fftri_f32):
+ * gst-libs/gst/fft/kiss_fftr_f32.h:
+ * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
+ (kiss_fftr_f64), (kiss_fftri_f64):
+ * gst-libs/gst/fft/kiss_fftr_f64.h:
+ * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
+ (kiss_fftr_s16), (kiss_fftri_s16):
+ * gst-libs/gst/fft/kiss_fftr_s16.h:
+ * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
+ (kiss_fftr_s32), (kiss_fftri_s32):
+ * gst-libs/gst/fft/kiss_fftr_s32.h:
+ * gst-libs/gst/fft/kiss_version:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ Add libgstfft, a FFT library based on Kiss FFT which is
+ BSD licensed. Supported sample formats are int16, int32,
+ float and double. For those formats a real FFT and IFFT
+ can be done, different windowing functions can be applied
+ and functions for extracting the magnitude and phase exist.
+ Fixes #468619.
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Integrate libgstfft into the docs.
+ * tests/check/Makefile.am:
+ * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
+ Add unit tests for libgstfft, currently only testing the FFT.
+ Unit tests for IFFT will follow soon.
+
+2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
+ (gst_sdp_message_init), (gst_sdp_message_uninit),
+ (is_multicast_address), (gst_sdp_message_as_text),
+ (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
+ (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
+ (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
+ (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
+ (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
+ (gst_sdp_media_init), (gst_sdp_media_uninit),
+ (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
+ (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
+ (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
+ (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
+ (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ Separate INIT_ARRAY() and related macros into two versions, one for
+ structures and one for pointers (e.g., INIT_ARRAY() and
+ INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
+ lists of emails and phone numbers.
+ Add missing const as appropriate.
+ Change all gint to guint since they all actually represent unsigned
+ values.
+ Do not use time as a variable name as it shadows the global time().
+ Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
+ Actually implement gst_sdp_message_add_time().
+ Make gst_sdp_message_add_time() take repeat times as an argument.
+ Store repeat times in GstSDPTime as a GArray rather than as gchar**.
+ Corrected the definition of gst_sdp_media_get_bandwidth() (was
+ misspelled as badwidth).
+ gst-indented and a little clean up. Fixes #471067.
+
+2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (volume_process_double), (volume_process_double_clamp),
+ (volume_process_float_clamp):
+ Correctly clamp float/double samples in the [-1.0,1.0] range to
+ prevent weird effects.
+ * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
+ Add unit tests for all samples types that had none before.
+
+2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ Need to include stdlib.h for abs() here too.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/law/alaw-decode.c: (alawdec_getcaps):
- * gst/law/mulaw-decode.c: (mulawdec_getcaps):
- Prevent warnings when negotiating caps (fixes #159338).
+ gst/playback/gststreaminfo.c: Fix build.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c:
+ Fix build.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- Remove old leftover that shouldn't be there...
+ gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c:
+ Clean up some half-disabled code and comment.
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_payload_audio_handle_event):
+ Return FALSE from the event handler to let the parent class handle the
+ event.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
+ Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ Bump the MTU to 1400.
+
+2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
+
+ gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
+ Original commit message from CVS:
+ 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
+ * gst/typefind/gsttypefindfunctions.c (plugin_init):
+ Add an audio/x-nsf typefind function for the nsfdec element.
+
+2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
+
+ gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ Included "myth://" on stream_uris list for enable buffering to mythtv files
+
+2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Fix parsing of RB blocks.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
+ (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
+ (gst_rtcp_unix_to_ntp):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Fix parsing of RB blocks.
+ Fix docs.
+ Added helper functions to convert to/from UNIX and NTP time.
+ API: gst_rtcp_ntp_to_unix()
+ API: gst_rtcp_unix_to_ntp()
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
+ (gst_rtp_buffer_get_header_len),
+ (gst_rtp_buffer_get_extension_data),
+ (gst_rtp_buffer_get_payload_subbuffer),
+ (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
+ (gst_rtp_buffer_ext_timestamp):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Fix some more docs.
+ Implement handling of packets with extensions.
+ Fix padding check in _validate().
+ Added function to get extension data.
+ API: gst_rtp_buffer_get_header_len()
+ API: gst_rtp_buffer_get_extension_data()
+
+2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Add some more docs for the queue-delay property and fix a typo in a
+ comment.
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ Fix typo.
+
+2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_change_state):
+ When skew slaving, try to hover around the middle of a segment so that
+ we at most drift by half a segment.
+ If we are aligning in the oposite direction of the clock skew, we don't
+ have to resync.
+
+2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Be less silly with the segment start, just apply the clock-base to the
+ timestamp.
+
+2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init),
+ (gst_base_rtp_depayload_finalize),
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_handle_sink_event),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Deprecate the queue handling thread thing and remove the code.
+ Use new method to calculate the extended timestamp.
+
+2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ (gst_rtcp_packet_sdes_copy_entry):
+ Use g_strndup which does exactly what we want.
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
+ (gst_rtp_buffer_ext_timestamp):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Add helper function to compare seqnums.
+ Add helper function to calculate extended timestamps.
+ API: gst_rtp_buffer_compare_seqnum()
+ API: gst_rtp_buffer_ext_timestamp()
+
+2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ (gst_rtcp_packet_sdes_get_entry),
+ (gst_rtcp_packet_sdes_copy_entry):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Fix and document SDES item data function.
+ Add new function that makes a proper copy of SDES item data.
+ API: gst_rtcp_packet_sdes_copy_entry()
+
+2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/Makefile.am:
+ The tcp and subparse plugins are under gst, but not totaly free of
+ dependencies. Handle selection inconfigure.ac, so that they show up
+ on the final list of what is build and what is not. Maybe they should
+ better be moved to ext.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
- Don't forward DISCONT events (fixes #159684).
+ Check if libxml provides HTML parser which subparse needs.
+ Original commit message from CVS:
+ Patch by: Daniel Díaz <yosoy@danieldiaz.org>
+ * configure.ac:
+ * gst/Makefile.am:
+ Check if libxml provides HTML parser which subparse needs.
+ Fixes #451970.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybin.c: (remove_sinks), (setup_sinks):
- Unlink manually since sometimes bin disposal (and therefore
- pad unlinking) is delayed, which will cause a new media file
- to not be able to start playing instantly.
+ ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c:
+ Fix typo and compilation on big endian systems.
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gststreaminfo.c: (stream_info_mute_pad):
- On mute of an unlinked stream, check for pad availability so
- we don't crash on unlinked pad.
+ gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
+ Original commit message from CVS:
+ * gst/subparse/gstssaparse.c:
+ Convert SSA newline codes into actual newline characters (#470766).
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_massage_index):
- Fix quite humiliating bug in omitting 0-sized index chunks but
- forgetting to count them for timestamps.
+ API: also add gst_install_plugins_supported() while we're at it (see #470456).
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/install-plugins.h:
+ * tests/check/libs/pbutils.c:
+ API: also add gst_install_plugins_supported() while we're at it
+ (see #470456).
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Actually leave the loop if we failed to sync. Don't crash.
+ API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ * gst-libs/gst/pbutils/missing-plugins.h:
+ * tests/check/libs/pbutils.c:
+ API: add gst_missing_*_installer_detail_new() convenience API so
+ that applications that know exactly what they're missing can request
+ installer detail strings for those items directly instead of having
+ to first create a dummy missing-plugin message and then get the
+ installer detail string from that. Fixes #470456.
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_get_audio_stream),
- (gst_dvd_demux_process_private):
- * gst/mpegstream/gstdvddemux.h:
- Fix crash (#159759). Doesn't work, though. :-(.
+ gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ We need to set up delayed-linking whenever the caps are non-fixed,
+ not just when there are multiple types - use gst_pad_is_fixed()
+ to test.
-2004-11-28 Benjamin Otte <otte@gnome.org>
+2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
- more overwriting protection due to modifying channels one by one
- instead of all at once
+ gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ (gst_missing_plugin_message_get_installer_detail):
+ Add missing separator in PID fallback case.
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
+ Original commit message from CVS:
+ * ext/alsa/Makefile.am:
+ There is no GST_PLUGINS_BASE_LIBS defined.
+ * ext/alsa/gstalsa.c:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
+ Add support for ALSA 24-bit formats.
+ snd_pcm_delay can return an error code, especially
+ during XRUNS. In that case, the best we can do is assume
+ delay = 0.
+ * gst/audioconvert/Makefile.am:
+ Add flags from -base before any more-remote dependencies.
+
+2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
+
+ gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
+ Original commit message from CVS:
+ Based on a patch by: Davyd <davyd at madeley dot id dot au>
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (volume_update_real_volume), (gst_volume_set_volume),
+ (gst_volume_init), (volume_process_int32),
+ (volume_process_int32_clamp), (volume_process_int24),
+ (volume_process_int24_clamp), (volume_process_int16),
+ (volume_process_int16_clamp), (volume_process_int8),
+ (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
+ * gst/volume/gstvolume.h:
+ Add support for int32, int24 and int8 to the volume element.
+ Fixes #445529.
+
+2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/examples/Makefile.am: Fix even more.
+ Original commit message from CVS:
+ * tests/examples/Makefile.am:
+ Fix even more.
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_normalize):
- Normalize using absolute values.
+2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ * gst-libs/gst/Makefile.am:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ * sys/v4l/v4lsrc_calls.c:
+ * tests/examples/Makefile.am:
+ * win32/common/config.h:
+ Revert unwanted commit. many thanks to moap. I want a fix for
+ https://thomas.apestaart.org/moap/trac/ticket/239
-2004-11-28 Julien MOUTTE <julien@moutte.net>
+2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+ * ChangeLog:
* configure.ac:
- * ext/Makefile.am:
- * ext/directfb/Makefile.am:
- * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_create),
- (gst_directfbvideosink_get_pixel_format),
- (gst_directfbvideosink_get_format_from_fourcc),
- (gst_directfbvideosink_fixate), (gst_directfbvideosink_getcaps),
- (gst_directfbvideosink_sink_link),
- (gst_directfbvideosink_change_state),
- (gst_directfbvideosink_chain), (gst_directfbvideosink_buffer_free),
- (gst_directfbvideosink_buffer_alloc),
- (gst_directfbvideosink_interface_supported),
- (gst_directfbvideosink_interface_init),
- (gst_directfbvideosink_navigation_send_event),
- (gst_directfbvideosink_navigation_init),
- (gst_directfbvideosink_set_property),
- (gst_directfbvideosink_get_property),
- (gst_directfbvideosink_finalize), (gst_directfbvideosink_init),
- (gst_directfbvideosink_base_init),
- (gst_directfbvideosink_class_init),
- (gst_directfbvideosink_get_type), (plugin_init):
- * ext/directfb/directfbvideosink.h: Adding a first version of
- directfbvideosink.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_init): Initializing some
- more.
-
-2004-11-28 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
- walk the samples backwards if out_channels > in_channels so we don't
- overwrite data
-
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/volume/gstvolume.c:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ * sys/v4l/v4lsrc_calls.c:
+ * tests/examples/Makefile.am:
+ * win32/common/config.h:
+ Original commit message from CVS: reviewed by: <delete if not using a buddy> patch by: <delete if not someone else's patch> * configure.ac: * docs/libs/Makefile.am: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * ext/gnomevfs/gstgnomevfssrc.c: * ext/gnomevfs/gstgnomevfssrc.h: * gst-libs/gst/Makefile.am: * gst-libs/gst/audio/gstaudiofilter.h: * gst/typefind/gsttypefindfunctions.c: * gst/volume/gstvolume.c: * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: * pkgconfig/gstreamer-plugins-base.pc.in: * sys/v4l/v4lsrc_calls.c: * tests/examples/Makefile.am: * win32/common/config.h:
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_link), (gst_audio_convert_change_state),
- (gst_audio_convert_channels):
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_unset_matrix),
- (gst_audio_convert_fill_identical),
- (gst_audio_convert_fill_compatible),
- (gst_audio_convert_detect_pos), (gst_audio_convert_fill_one_other),
- (gst_audio_convert_fill_others),
- (gst_audio_convert_fill_normalize),
- (gst_audio_convert_fill_matrix), (gst_audio_convert_setup_matrix),
- (gst_audio_convert_passthrough), (gst_audio_convert_mix):
- * gst/audioconvert/gstchannelmix.h:
- Implement a channel mixer.
+2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
+ gst-libs/gst/audio/audio.c: Clarify the docs a little.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Clarify the docs a little.
- * ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsa.c (gst_alsa_set_clock):
- Make alsasink actually honor gst_element_set_clock and use that
- clock instead of its internal one.
-
-2004-11-27 Christophe Fergeau <teuf@gnome.org>
-
- * gst/playback/gstplaybasebin.c: (setup_source): fixed a caps leak
- (gst_play_base_bin_change_state): nullify source and decoder when
- going from READY to NULL so that we don't try to do weird stuff with
- them when going from NULL to READY
- * gst/playback/gstplaybin.c: (gst_play_bin_init): use gst_object_unref
- instead of g_object_unref
- (gen_video_element), (gen_audio_element): more refcounting fixes, now
- it should be correct
- (gst_play_bin_change_state): don't call remove_sinks if we are
- currently disposing the object
-
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Don't forget bass if it's there. Else left channel is silent...
-
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop),
- (gst_a52dec_change_state):
- Don't do sample adjusting anymore, we use float audio now.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
- Don't fixate to non-existing properties.
-
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_channels),
- (gst_a52dec_change_state):
- Advertise that we can do surround sound.
-
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_reneg):
- Add buffer-frames=0.
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get_type),
- (dvdreadsrc_init), (dvdreadsrc_get_event_mask),
- (dvdreadsrc_get_query_types), (dvdreadsrc_get_formats),
- (dvdreadsrc_srcpad_event), (dvdreadsrc_srcpad_query),
- (_seek_title), (_seek_chapter), (get_next_cell_for), (_read),
- (seek_sector), (dvdreadsrc_get), (dvdreadsrc_open_file),
- (dvdreadsrc_change_state), (dvdreadsrc_uri_get_type),
- (dvdreadsrc_uri_get_protocols), (dvdreadsrc_uri_get_uri),
- (dvdreadsrc_uri_set_uri), (dvdreadsrc_uri_handler_init):
- * ext/dvdread/dvdreadsrc.h:
- Add seeking, querying for bytes, sectors, title, angle and
- chapter. Handle multiple chapters. Relicense to LGPL because
- Billy agreed on that (thanks Billy!).
-
-2004-11-27 Christophe Fergeau <teuf@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
- call parent dispose method
-
-2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
-
- * gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
- (gst_audio_clock_get_internal_time):
- Fix active <-> inactive transitions: ensure time value always
- grows and avoid abrupt value changes.
-
-2004-11-27 Arwed v. Merkatz <v.merkatz@gmx.net>
+2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- * gst/tta/Makefile.am:
- * gst/tta/crc32.h:
- * gst/tta/filters.h:
- * gst/tta/gsttta.c:
- * gst/tta/gstttadec.c:
- * gst/tta/gstttadec.h:
- * gst/tta/gstttaparse.c:
- * gst/tta/gstttaparse.h:
- * gst/tta/ttadec.h:
- added TTA parser and decoder
-
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (probe_triggered), (check_queue), (buffer_underrun),
- (buffer_running), (buffer_overrun), (gen_source_element),
- (setup_source):
- * gst/playback/gstplaybasebin.h:
- Implement buffering. Needs some more work.
+ gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c:
+ Enable liboil for float and add more details about problems with
+ int16.
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
+ Original commit message from CVS:
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
+ Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
+
+2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
+ When calculating the first timestamp of the buffers, don't go below 0
+ and clip the samples because the offset was on the eos page.
+ Fixes #466717.
+
+2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
+ (gst_ogg_demux_collect_chain_info):
+ Also submit the eos page when trying to find the first timestamp.
+ See #466717.
+
+2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h:
+ Use gst_util_uint64_scale() instead of doing the math
+ with double for GST_FRAMES_TO_CLOCK_TIME() and
+ GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
+ prevents rounding errors. Fixes #467667.
+
+2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
+ (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ Small cleanups.
+ On shutdown, don't read the control socket yet.
+ Set timeout value correctly in all cases.
+ Add function to check if the server accepts reads or writes.
+ API: gst_rtsp_connection_poll()
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ Fix compilation with -pedantic.
+ Add enum for _poll.
+
+2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
+ Override the preroll vmethod instead of overriding the render method
+ twice.
+
+2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
+
+ gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester at tester ca>
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
+ (gst_basertppayload_getcaps):
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Add getcaps vfunc to basertppayload. See #465146.
+
+2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
+ Only post buffering messages when we are a stream.
+
+2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/pbutils/: Small docs fix and addition.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ Small docs fix and addition.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Fix ilog mask range overflow.
+2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.c: Don't use new API.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
+ Don't use new API.
+
+2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.*: Make love to appsink.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
+ (gst_app_sink_class_init), (gst_app_sink_dispose),
+ (gst_app_sink_flush_unlocked), (gst_app_sink_start),
+ (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
+ (gst_app_sink_render), (gst_app_sink_get_caps),
+ (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
+ (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
+ * gst-libs/gst/app/gstappsink.h:
+ Make love to appsink.
+ Make it support pulling of the preroll buffer.
+ Add docs and debug statements.
+ Fix some races wrt to EOS handling and stopping.
+ Implement getcaps.
+ Implement FLUSHING.
+ API: gst_app_sink_pull_preroll()
+
+2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/icles/: Add a dumb little test for textoverlay alignments.
+ Original commit message from CVS:
+ * tests/icles/.cvsignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-textoverlay.c:
+ Add a dumb little test for textoverlay alignments.
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps):
- Don't omit the last (which in case of dmix is the only :) )
- channel count. Don't set channels if <= 2.
+ ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
+ Original commit message from CVS:
+ Patch by: Dan Williams <dcbw redhat com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ API: add "line-alignment" property (#459334). Add gtk-doc blurb for
+ "silent" property so there's a Since tag in the API reference.
-2004-11-26 Christophe Fergeau <teuf@gnome.org>
+2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element): Removed 2 obsolete comments
+ * ChangeLog:
+ fix ... by: lines
+ Original commit message from CVS:
+ fix ... by: lines
+
+2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_set_outcaps):
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Improve caps negotiation so that downstream elements can confiure
+ certain RTP properties by fixing them on the caps. See #465146.
+ Add docs.
+
+2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Mark as deprecated some macros which were presumably meant to be
+ private API and accidentally exposed in the public header file.
+ Also actually _init() lock (only works at the moment because the
+ struct is zeroed out when created and the initial values in the
+ mutex struct are zeroes too). (#459585)
-2004-11-26 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/vorbis/oggvorbisenc.c
- * ext/vorbis/vorbisenc.c :
- change description fields of those plugins to differentiate them
- (pitivi show Encoders by description, they had the same one)
+ docs/libs/Makefile.am: Remove cruft and do some cleanups.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ Remove cruft and do some cleanups.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ Prepare for comming gtkdoc features (rebase against online docs).
-2004-11-25 Christophe Fergeau <teuf@gnome.org>
+2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiorate/gstaudiorate.c: Debug output fixes.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Debug output fixes.
+ * tests/check/elements/audiorate.c: (do_perfect_stream_test),
+ (GST_START_TEST):
+ Change the number of buffers used; 500 is too many and leads to
+ timeouts.
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
- (gst_play_bin_set_property), (gen_video_element),
- (gen_audio_element):
- Refcounting fixes for provided audio-/videosinks.
+2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Printf format fixes (#465028).
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c:
+ * gst/videorate/gstvideorate.c:
+ Printf format fixes (#465028).
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element), (setup_sinks), (gst_play_bin_change_state):
- Don't reference all sinks, but only the video- and audiosinks.
- The vis. element should be disposed when we're done with it.
- We don't have any reason to keep it around. This fixes warnings
- when reusing playbin for playing multiple audio files with
- vis. enabled. Also release audio device on pause - idea stolen
- from Rhythmbox.
+2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ If we have a large (> 1 second) discontinuity, push a series of
+ smaller buffers rather than a single very large buffer. Avoids
+ unreasonably large single buffer allocations when encountering a
+ large gap.
+ * tests/check/elements/audiorate.c: (GST_START_TEST),
+ (audiorate_suite):
+ Add a test for this.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
- (gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps):
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
- (gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
- (gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
- (gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
- (gst_faad_change_state), (plugin_init):
- * ext/faad/gstfaad.h:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/audio.c: (plugin_init):
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_get_channel_positions),
- (gst_audio_set_channel_positions),
- (gst_audio_set_structure_channel_positions_list),
- (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
- (gst_audio_fixate_channel_positions):
- * gst-libs/gst/audio/multichannel.h:
- * gst-libs/gst/audio/testchannels.c: (main):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_init),
- (gst_audio_convert_dispose), (gst_audio_convert_getcaps),
- (gst_audio_convert_parse_caps), (gst_audio_convert_link),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- * gst/audioconvert/plugin.c: (plugin_init):
- Surround sound support.
+2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Fixes: #465015
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_remove_probe), (queue_threshold_reached):
+ Patch by: Josep Torra Valles <josep@fluendo.com>
+ Fixes: #465015
+ Make sure we remove the check_queues buffer probe from the
+ correct queue to avoid racily going back to "buffering 99%" when
+ buffering is actually complete.
+ Also, fix the spelling of Josep's surname in the ChangeLog.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
- Fix position for discont if we're close as well. Nitpicking, but
- saves a few milliseconds of extra waiting or skipping.
+2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/ogg/gstoggmux.c: Do not leak oggmux instance.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Do not leak oggmux instance.
+ * ext/vorbis/vorbisenc.c:
+ Also log values.
- * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter):
- We sometimes need parsers for playback, so add those too.
+2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/: Updated translations.
+ Original commit message from CVS:
+ * po/hu.po:
+ * po/it.po:
+ * po/nl.po:
+ * po/uk.po:
+ * po/vi.po:
+ Updated translations.
- * configure.ac:
- * gst/apetag/Makefile.am:
- * gst/apetag/apedemux.c: (gst_ape_demux_get_type),
- (gst_ape_demux_base_init), (gst_ape_demux_class_init),
- (gst_ape_demux_init), (gst_ape_demux_get_src_formats),
- (gst_ape_demux_get_src_query_types),
- (gst_ape_demux_handle_src_query), (gst_ape_demux_get_event_mask),
- (gst_ape_demux_handle_src_event), (gst_ape_demux_handle_event),
- (gst_ape_demux_typefind_peek), (gst_ape_demux_typefind_get_length),
- (gst_ape_demux_typefind_suggest), (gst_ape_demux_typefind),
- (gst_ape_demux_parse_tags), (gst_ape_demux_stream_init),
- (gst_ape_demux_stream_data), (gst_ape_demux_loop),
- (gst_ape_demux_change_state):
- * gst/apetag/apedemux.h:
- * gst/apetag/apetag.c: (plugin_init):
- * gst/typefind/gsttypefindfunctions.c: (apetag_type_find),
- (plugin_init):
- APE v1/2 tag reader plus typefind function.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
- * configure.ac:
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- Remove hacks for older core. Require newer core version
- accordingly.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/cdxaparse/Makefile.am:
- * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_get_type),
- (gst_cdxaparse_class_init), (gst_cdxaparse_init),
- (gst_cdxaparse_loop), (gst_cdxaparse_change_state), (plugin_init):
- * gst/cdxaparse/gstcdxaparse.h:
- * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_get_type),
- (gst_cdxastrip_base_init), (gst_cdxastrip_class_init),
- (gst_cdxastrip_init), (gst_cdxastrip_get_src_formats),
- (gst_cdxastrip_get_src_query_types),
- (gst_cdxastrip_handle_src_query), (gst_cdxastrip_get_event_mask),
- (gst_cdxastrip_handle_src_event), (gst_cdxastrip_strip),
- (gst_cdxastrip_sync), (gst_cdxastrip_handle_event),
- (gst_cdxastrip_chain), (gst_cdxastrip_change_state):
- * gst/cdxaparse/gstcdxastrip.h:
- SVCD/VCD header stripping separated from CDXA image parsing.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (plugin_init):
- Add VCD/SVCD header typefinding for VCD/SVCD.
- * sys/vcd/vcdsrc.c: (gst_vcdsrc_get_type), (gst_vcdsrc_base_init),
- (gst_vcdsrc_class_init), (gst_vcdsrc_init),
- (gst_vcdsrc_set_property), (gst_vcdsrc_get_property),
- (gst_vcdsrc_get_event_mask), (gst_vcdsrc_get_query_types),
- (gst_vcdsrc_get_formats), (gst_vcdsrc_srcpad_event),
- (gst_vcdsrc_srcpad_query), (gst_vcdsrc_get),
- (gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
- (gst_vcdsrc_change_state), (gst_vcdsrc_msf),
- (gst_vcdsrc_recalculate), (gst_vcdsrc_uri_get_type),
- (gst_vcdsrc_uri_get_protocols), (gst_vcdsrc_uri_get_uri),
- (gst_vcdsrc_uri_set_uri), (gst_vcdsrc_uri_handler_init):
- * sys/vcd/vcdsrc.h:
- Fix up, add seeking, querying, URI interface. Works in totem now.
-
-2004-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
+ Original commit message from CVS:
+ patch by: Yang Hong <hongyang@redflag-linux.com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ Add 'silent' property to GstTimeOverlay. Fixes #462979
- * configure.ac:
- back to CVS
+2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
+
+ Add connection-speed property. Fixes #464690.
+ Original commit message from CVS:
+ Patch by: Josep Torre Valles <josep@fluendo.com>
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
+ (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
+ (gst_uri_decode_bin_get_property), (gen_source_element):
+ Add connection-speed property. Fixes #464690.
+
+2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
+
+ Fix compilation on windows. Fixes #464320.
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * configure.ac:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect):
+ Fix compilation on windows. Fixes #464320.
+
+2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
+
+ gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
+ Original commit message from CVS:
+ Patch by: Josep Torre Valles <josep@fluendo.com>
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_init), (queue_threshold_reached),
+ (gen_source_element), (setup_substreams),
+ (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
+ (gst_play_base_bin_get_streaminfo_value_array):
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_property), (gst_play_bin_get_property),
+ (gst_play_bin_handle_redirect_message):
+ Move connection-speed property from playbin to playbasebin so that we
+ can also configure it in source elements that have the connection-speed
+ property. Fixes #464028.
+ Add some debug info here and there.
+
+2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
+ Properly respond to conversion queries. Fixes #464079.
+
+2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
+ (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
+ (gst_audio_test_src_init_sine_table),
+ (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Add float/double and int32 support to audiotestsrc. Fixes #460422.
+ Also set the default volume to the default value specified in the
+ GParamSpec.
+
+2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
+
+ gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
+ Original commit message from CVS:
+ Patch by: Jens Granseuer <jensgr at gmx dot net>
+ * gst/audioconvert/gstaudioquantize.c:
+ Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
+
+2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
+ Add rdt manager for rdt transport.
+ Fix parsing of RDT transport.
+
+2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
-=== release 0.8.6 ===
+=== release 0.10.14 ===
-2004-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ Release 0.10.14
+ Original commit message from CVS:
+ Release 0.10.14
+
+2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
* po/af.po:
* po/az.po:
* po/cs.po:
+ * po/de.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
- releasing 0.8.6, "IOU Love"
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
+ Original commit message from CVS:
+ * tests/check/libs/audio.c: (GST_START_TEST):
+ Fix the test to reflect the behaviour of gst_audio_clip_buffer.
+
+2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ When clipping a buffer with no timestamp, assume it is
+ within the segment without warnings.
+ Fixes: #460978
+
+2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
+ Fire the signal on the object, not the interface.
+
+2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/.cvsignore:
+ Ber. Don't include the full path, idiot.
+
+2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/.cvsignore:
+ Ignore generated files.
+
+2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/interfaces-marshal.list:
+ * gst-libs/gst/interfaces/rtspextension.c:
+ * gst-libs/gst/interfaces/rtspextension.h:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtsp.h:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
+ (gst_rtsp_extension_detect_server),
+ (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
+ (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
+ (gst_rtsp_extension_configure_stream),
+ (gst_rtsp_extension_get_transports),
+ (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
+ * gst-libs/gst/rtsp/gstrtspextension.h:
+ * gst-libs/gst/rtsp/rtsp-marshal.list:
+ Move the rtspextension.h interface into gstrtspextension.h
+ as part of libgstrtsp instead of libgstinterfaces, because it's
+ only for use within plugins, not applications.
+ Add stuff to do the enum & marshal generation needed in libgstrtsp now.
+ Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
+ signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
+ is abstract.
+
+2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/interfaces/: Fix marshaller for the send signal.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/interfaces-marshal.list:
+ * gst-libs/gst/interfaces/rtspextension.c:
+ (gst_rtsp_extension_iface_init),
+ (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
+ * gst-libs/gst/interfaces/rtspextension.h:
+ Fix marshaller for the send signal.
+ Add URL to stream selection interface method.
+
+2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/Makefile.am:
+ Pull in our dependencies from -base before those from outside.
+
+2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ API: gst_rtsp_base64_decode_ip()
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
+ * gst-libs/gst/rtsp/gstrtspbase64.h:
+ API: gst_rtsp_base64_decode_ip()
+ Added function to decode Base64 in-place.
+
+2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/libs/.cvsignore: Ignore the mixer test binary.
+ Original commit message from CVS:
+ * tests/check/libs/.cvsignore:
+ Ignore the mixer test binary.
+
+2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
+ Gratuitous comment change to trigger a rebuild on the buildbots.
+
+2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
+ Original commit message from CVS:
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
+ (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
+ (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
+ (gst_sdp_media_get_format), (gst_sdp_media_get_information),
+ (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
+ (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
+ (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
+ (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
+ (gst_sdp_media_get_attribute_val):
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ Constify args where we can.
+
+2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/rtspextension.c:
+ (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
+ (gst_rtsp_extension_detect_server),
+ (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
+ (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
+ (gst_rtsp_extension_configure_stream),
+ (gst_rtsp_extension_get_transports),
+ (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
+ * gst-libs/gst/interfaces/rtspextension.h:
+ Move interface for RTSP extensions from -good to here.
+ Added helper methods to invoke interface methods.
+
+2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Fix some more RTSP docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
+ (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
+ (gst_rtsp_message_init_response),
+ (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
+ (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
+ (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
+ (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
+ (gst_rtsp_message_get_body), (dump_key_value):
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
+ (parse_npt_range), (parse_clock_range), (parse_smpte_range),
+ (gst_rtsp_range_parse):
+ * gst-libs/gst/rtsp/gstrtsprange.h:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ Fix some more RTSP docs.
+ Add some missing methods for dealing with messages.
+
+2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Added beginnings of RTSP documentation.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
+ * gst-libs/gst/rtsp/gstrtspbase64.h:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect), (add_auth_header),
+ (gst_rtsp_connection_write), (gst_rtsp_connection_send),
+ (read_body), (gst_rtsp_connection_receive),
+ (gst_rtsp_connection_next_timeout),
+ (gst_rtsp_connection_reset_timeout),
+ (gst_rtsp_connection_set_auth):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
+ (parse_npt_range), (parse_clock_range), (parse_smpte_range),
+ (gst_rtsp_range_parse):
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ Added beginnings of RTSP documentation.
+
+2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Document the SDP library.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/sdp/gstsdp.h:
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
+ (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
+ (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
+ (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
+ (gst_sdp_message_get_attribute_val),
+ (gst_sdp_message_add_attribute), (gst_sdp_media_new),
+ (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
+ (gst_sdp_media_get_media), (gst_sdp_media_set_media),
+ (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
+ (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
+ (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
+ (gst_sdp_media_get_format), (gst_sdp_media_add_format),
+ (gst_sdp_media_get_information), (gst_sdp_media_set_information),
+ (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
+ (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
+ (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
+ (gst_sdp_media_set_key), (gst_sdp_media_get_key),
+ (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
+ (gst_sdp_media_get_attribute_val_n),
+ (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
+ (print_media), (gst_sdp_message_dump):
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ Document the SDP library.
+ Add some of the missing SDPMedia methods.
+
+2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Move SDP and RTSP from helper objects in -good to a reusable library.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
+ * gst-libs/gst/rtsp/gstrtspbase64.h:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
+ (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
+ (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
+ (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
+ (parse_response_status), (parse_request_line), (parse_line),
+ (gst_rtsp_connection_read), (read_body),
+ (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
+ (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
+ (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
+ (gst_rtsp_connection_set_auth):
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
+ (gst_rtsp_strresult), (gst_rtsp_method_as_text),
+ (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
+ (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
+ (gst_rtsp_find_method):
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
+ (gst_rtsp_message_new), (gst_rtsp_message_init),
+ (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
+ (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
+ (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
+ (gst_rtsp_message_free), (gst_rtsp_message_add_header),
+ (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
+ (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
+ (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
+ (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
+ (gst_rtsp_message_dump):
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
+ (parse_npt_range), (parse_clock_range), (parse_smpte_range),
+ (gst_rtsp_range_parse), (gst_rtsp_range_free):
+ * gst-libs/gst/rtsp/gstrtsprange.h:
+ * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
+ (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
+ (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
+ (range_as_text), (rtsp_transport_mode_as_text),
+ (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
+ (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
+ (gst_rtsp_transport_free):
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
+ (gst_rtsp_url_free), (gst_rtsp_url_set_port),
+ (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/sdp/gstsdp.h:
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
+ (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
+ (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
+ (gst_sdp_attribute_init), (gst_sdp_message_new),
+ (gst_sdp_message_init), (gst_sdp_message_uninit),
+ (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
+ (gst_sdp_media_uninit), (gst_sdp_media_free),
+ (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
+ (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
+ (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
+ (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
+ (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
+ (gst_sdp_message_get_attribute_val),
+ (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
+ (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
+ (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
+ (gst_sdp_media_get_attribute_val_n),
+ (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
+ (read_string), (read_string_del), (gst_sdp_parse_line),
+ (gst_sdp_message_parse_buffer), (print_media),
+ (gst_sdp_message_dump):
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ Move SDP and RTSP from helper objects in -good to a reusable library.
+ Use a proper gst_ namespace.
+
+2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
+ (vorbis_dec_flush_decode):
+ Use the new buffer clipping function from gstaudio here.
+
+2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ API: Add buffer clipping function for raw audio buffers. Fixes #456656.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
+ * gst-libs/gst/audio/audio.h:
+ * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
+ API: Add buffer clipping function for raw audio buffers. Fixes #456656.
+ Also add deprecation guards for gst_audio_structure_set_int() to the
+ header.
+
+2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Cleanup the docs.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
+ Original commit message from CVS:
+ Patch by: Dan Williams <dcbw at redhat dot com>
+ * gst/playback/gstplaybasebin.c:
+ (gst_play_base_bin_get_streaminfo_value_array):
+ Don't return NULL when querying the stream info value array but instead
+ return an empty array. Fixes #459204.
- * gst/playback/gstplaybasebin.c:
- Fix unplayable files error handling. Fixes #158365
+2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gsturidecodebin.c: Init debug category before using it.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c:
+ Init debug category before using it.
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixer.h:
+ Add padding vars in place of the signal pointers
+ when building with DISABLE_DEPRECATED so that the
+ interface structure doesn't change size.
+
+2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
+
+ Fixes: #152864
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsamixer.h:
+ * ext/alsa/gstalsamixerelement.c:
+ * ext/alsa/gstalsamixertrack.c:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ * tests/check/Makefile.am:
+ * tests/check/libs/mixer.c:
+ Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ Fixes: #152864
+ Add support for notifying mixer changes on the message bus, and
+ implement it in alsamixer.
+ API: gst_mixer_get_mixer_flags
+ API: gst_mixer_message_parse_mute_toggled
+ API: gst_mixer_message_parse_record_toggled
+ API: gst_mixer_message_parse_volume_changed
+ API: gst_mixer_message_parse_option_changed
+ API: GstMixerMessageType
+ API: GstMixerFlags
+
+2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
+
+ sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
+ xcontext->im_format is only for testing XShm support (as the header
+ file comments document). Use xvimage->im_format for everything else.
+ Avoids spurious warnings on buffer allocation before setcaps.
- * gst/typefind/gsttypefindfunctions.c:
- Fix broken mp3 typefinding. Fixes #158375
+2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ tests/: We should use $(LIBM).
+ Original commit message from CVS:
+ * tests/examples/volume/Makefile.am:
+ * tests/icles/Makefile.am:
+ We should use $(LIBM).
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c:
- Fix sync on broken files. Fixes #158976
+ tests/icles/Makefile.am: This needs -lm.
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ This needs -lm.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add stdlib include (free, atoi, exit).
+ Original commit message from CVS:
+ * examples/app/appsrc_ex.c:
+ * examples/switch/switcher.c:
+ * ext/neon/gstneonhttpsrc.c:
+ * ext/timidity/gstwildmidi.c:
+ * ext/x264/gstx264enc.c:
+ * gst/mve/mveaudioenc.c: (mve_compress_audio):
+ * gst/rtpmanager/gstrtpclient.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/spectrum/demo-audiotest.c:
+ * gst/spectrum/demo-osssrc.c:
+ * sys/dvb/gstdvbsrc.c:
+ Add stdlib include (free, atoi, exit).
+
+2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init), (gst_basertppayload_init),
+ (gst_basertppayload_set_property),
+ (gst_basertppayload_get_property):
+ Don't break ABI, restore previous ranges. Keep the default random
+ selection of timestamp and seqnum offset but as soon as the app sets a
+ specific value, use that one.
+
+2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
+
+ sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
+ Original commit message from CVS:
+ Patch by: Bastien Nocera <hadess at hadess dot net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
+ (gst_xvimagesink_init), (gst_xvimagesink_class_init):
+ * sys/xvimage/xvimagesink.h:
+ Add option to turn off double-buffering for debugging purposes.
+ Fixes #437169.
+
+2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
+
+ sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
+ Original commit message from CVS:
+ Patch by: Jorn Baayen <jorn at openedhand dot com>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
+ (gst_ximagesink_set_property), (gst_ximagesink_get_property),
+ (gst_ximagesink_init), (gst_ximagesink_class_init):
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
+ (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
+ (gst_xvimagesink_init), (gst_xvimagesink_class_init):
+ * sys/xvimage/xvimagesink.h:
+ add 'handle-expose' property. Useful for video widgets which may want to
+ be in control of Expose behaviour. Fixes #380625
+
+2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init), (gst_basertppayload_init),
+ (gst_basertppayload_event), (gst_basertppayload_push),
+ (gst_basertppayload_set_property),
+ (gst_basertppayload_get_property),
+ (gst_basertppayload_change_state):
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Fix ranges of rtp payloader properties so that the full range can be
+ used in addition to -1 (random).
+ Fix wrong seqnum reporting in caps.
+ Fixes #420326.
+
+2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videorate/gstvideorate.c: Use boilerplate.
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_init),
+ (gst_video_rate_query):
+ Use boilerplate.
+ Add latency query, might not be perfect yet but already works a lot
+ better. Fixes #442557.
+
+2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_setcaps):
+ * sys/xvimage/xvimagesink.h:
+ After a caps change, redraw our borders to avoid garbage left there
+ when the image format changes to a smaller size, like 16:9 -> 4:3
+ Also, hold the flow_lock a bit longer in the set_caps while we're
+ fiddling with the xcontext.
+
+2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
+ Original commit message from CVS:
+ * Makefile.am:
+ * configure.ac:
+ * tests/Makefile.am:
+ Remove bogus check for libcheck, since we check for
+ gstreamer-check and it pulls in the required info from there, and we
+ weren't actually _using_ the information for libcheck ourselves
+ anyway.
+
+2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_caps_to_pixfmt):
+ Fix the r_mask test for RGBA32 on little-endian.
+ Fix a stupid typo that would have obviously broken
+ compilation on big-endian, if anyone was testing.
+
+2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
+ (paint_hline_str4):
+ * gst/videotestsrc/videotestsrc.h:
+ Add alpha to the color struct.
+ Use a default alpha value of 255 instead of 128.
+
+2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (no_more_pads_full),
+ (setup_source):
+ Clear the dynamic pads counter when starting a new uri. This makes
+ reusing playbin work again.
+ Fixes #454264.
+
+2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ configure.ac: Use pkg-config to locate check.
+ Original commit message from CVS:
+ * configure.ac:
+ Use pkg-config to locate check.
+
+2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Fix 'make check' build against core CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ Fix 'make check' build against core CVS.
+
+2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/: Make gtk-doc happy.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Make gtk-doc happy.
- patch by: Edward Hervey <bilboed@bilboed.com>
+2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/libpng/gstpngenc.c:
- Copy over buffer properties. Fixes #158832
+ gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_callback):
+ Quick hack to make audiosinks stop at EOS when operating in
+ pull-mode; needs to be fixed properly some day.
+
+2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Fix location of includes in the docs.
+
+2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
+ (img_get_alpha_info):
+ Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
+ of the existing BGRA32 and RGBA32 formats with the alpha at the other
+ end of the word. Partially fixes #451908
+
+2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/: Simplify --extra-dir as gtkdoc scans recursively.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Simplify --extra-dir as gtkdoc scans recursively.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
+ (gst_adder_request_new_pad):
+ Make getcaps more robust by not using the proxycaps function. This makes
+ sure that we don't end up recursively calling getcaps upstream.
+ See #316248.
+
+2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c:
+ Include math.h to fix compilation.
+
+2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
+ Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
+ format, as produced by some dc1394 cameras like the iSight.
+ See http://www.fourcc.org/yuv.php#IYU1
+
+2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
+ (check_default), (audio_convert_prepare_context),
+ (audio_convert_clean_context), (audio_convert_convert):
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_dithering_get_type),
+ (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
+ (gst_audio_convert_init), (gst_audio_convert_set_caps),
+ (gst_audio_convert_set_property), (gst_audio_convert_get_property):
+ * gst/audioconvert/gstaudioconvert.h:
+ * gst/audioconvert/gstaudioquantize.c:
+ (gst_audio_quantize_setup_noise_shaping),
+ (gst_audio_quantize_free_noise_shaping),
+ (gst_audio_quantize_setup_dither),
+ (gst_audio_quantize_free_dither),
+ (gst_audio_quantize_setup_quantize_func),
+ (gst_audio_quantize_setup), (gst_audio_quantize_free):
+ * gst/audioconvert/gstaudioquantize.h:
+ Implement dithering and noise shaping in audioconvert. By default now
+ TPDF dithering (and no noise shaping) will be used when converting
+ from a higher bit depth to 20 bit depth or smaller, otherwise
+ everything will be as it is now.
+ For the last audioconvert in a pipeline it would make sense to
+ use some kind of noise shaping, enabling it by default for all
+ conversions would give undesired results though. Fixes #360246.
+ * tests/check/elements/audioconvert.c: (setup_audioconvert),
+ (GST_START_TEST):
+ Adjust unit test for the new audioconvert.
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/dvdread/dvdreadsrc.c:
- Fixes invalid reads (#158462)
+ gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
+ Use other metrics as well when estimating the buffer level.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- Probe less and cache it. Fixes #159187.
+ gst/playback/gstplaybasebin.c: Small debug improvement.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
+ Small debug improvement.
+ * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
+ (plugin_init):
+ Tweak the rate estimation period.
+ When calculating the buffer filledness in rate estimation mode, don't
+ mix it with other metrics.
+
+2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
+ (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
+ When creating the groups, allow for a 5 second, unlimited buffers
+ preroll phase after which we expose the group.
+ When the group is exposed, use a small number of buffers up to a 2
+ second limit. Also disconnect the overrun signal from multiqueue when we
+ exposed the group because it is not needed anymore.
+
+2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
+ Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
+ to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
+ (#451707); also, output some debugging info when dealing with
+ freeform strings.
+ * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
+ Add unit test for the above.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videorate/gstvideorate.c:
- Handle all video formats. Fixes #159186.
-
-2004-11-16 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/synaesthesia/gstsynaesthesia.c:
- (gst_synaesthesia_class_init), (gst_synaesthesia_init),
- (gst_synaesthesia_dispose), (gst_synaesthesia_finalize),
- (gst_synaesthesia_sink_link), (gst_synaesthesia_src_getcaps),
- (gst_synaesthesia_src_link), (gst_synaesthesia_chain),
- (gst_synaesthesia_change_state), (plugin_init):
- Fix up synaesthesia to work under different samplerates/ buffer sizes.
- Force 320x200 output, as that's the only thing the underlying
- synaesthesia implementation supports. Still needs to be made
- re-entrant.
-
-2004-11-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
+ Add description for Windows Media RTP caps.
+ * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
+ Remove RTP fields that don't define the format from caps.
- * configure.ac:
- Fix mpeg2enc configure check (similar to mplex check below).
+2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-14 Koop Mast <kwm@rainbow-runner.nl>
+ ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
+ Skip empty buffers, but not empty header buffers. That way the original
+ vorbisdec unit test still passes (#451145); also, take into account
+ that those empty packets might carry a granulepos.
+ * tests/check/Makefile.am:
+ * tests/check/elements/vorbisdec.c:
+ (_create_codebook_header_buffer), (_create_audio_buffer),
+ (GST_START_TEST), (vorbisdec_suite):
+ Add unit test that sends an empty packet.
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
+ Don't error out on 0-sized packets, just emit a warning because this is
+ not a fatal error. Fixes #451145.
+
+2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/plugins/: Update docs with caps info.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Update docs with caps info.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- Fix for gcc-2.95 (fixes #158221).
+2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/POTFILES.in: Add more files with translatable strings (#450875).
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Add more files with translatable strings (#450875).
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- Re-add clock distribution hack (until new core is released).
- Fixes #158125.
+2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-13 Arwed v. Merkatz <v.merkatz@gmx.net>
- * configure.ac:
- fix mplex configure check segfaulting on some systems (bug #140994)
+ ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
+ The chain should be freed if we error out here, else it will leak.
+ * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
+ (cleanup_decodebin):
+ Don't forget to *properly* remove the signals, else it will leak.
-2004-11-13 Benjamin Otte <otte@gnome.org>
+2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
- add debugging
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- do a wait when we enter the loop func with no data available to
- write instead of getting into an 100% CPU loop by just returning and
- being called again by the scheduler
+ MAINTAINERS: Updating all the maintainers files
+ Original commit message from CVS:
+ * MAINTAINERS:
+ Updating all the maintainers files
-2004-11-13 Jan Schmidt <thaytan@mad.scientist.com>
+2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_get_type),
- (libvisual_log_handler), (gst_visual_getcaps),
- (gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
- (plugin_init):
- Update libvisual to 0.1.7. Link in the debug handling to gstreamer
- * ext/smoothwave/Makefile.am:
- * ext/smoothwave/demo-osssrc.c: (main):
- * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
- (gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
- (gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
- (plugin_init):
- * ext/smoothwave/gstsmoothwave.h:
- Make gstsmoothwave a working element in the 20th century.
-
- * gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
- Fix incorrect link function
-
-2004-11-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
+ (main):
+ Destroy and recreate parse-launch based pipeline after stop to be able
+ to play again. Reorder some code and add more comments.
- * gst/volume/gstvolume.c:
- Allow buffer-frames=0.
+2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
-2004-11-12 Iain <iaingnome@gmail.com>
+ gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad):
+ When handling a delayed-caps notification case, mark
+ the group as dynamic so that the nbdynamic count is
+ incremented and decremented correctly. Fixes: #449156
+ Patch by: Wim Taymans <wim@fluendo.com>
- * configure.ac: Check for polypaudio
+2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
- * ext/Makefile.am: Build the polyp dir
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * win32/common/config.h:
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ 2007-06-19 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_init): Enable pull-mode operation.
- * ext/polyp: The polypsink sources.
+2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
-2004-10-30 Iain <iaingnome@gmail.com>
+ gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Change minimum rate back to 1000 to allow low-sample-rate wav files
+ to play back.
- * gst/interleave/interleave.c (interleave_unlink): Change the src pads
- caps to reflect the new number of channels.
+2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/vi.po: Update translations.
+ Original commit message from CVS:
+ * po/vi.po:
+ Update translations.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Fix for negotiation order problem. This would show when the
- ALSA loopfuction was called before any other function. ALSA
- wouldn't do anything because we're not negotiated yet, leading
- to an infinite loop. Showed in e.g. Rhythmbox. Fixes #158006.
+2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
-2004-11-11 Tim-Philipp Müller <t.i.m@zen.co.uk>
+ gst/playback/gstqueue2.c: Fix compile error from ignored return value.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c:
+ Fix compile error from ignored return value.
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- No warnings (#157986).
+ gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
+ Original commit message from CVS:
+ * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
+ Update tmpbuf for all neccesary rows, not just one, as is required
+ when downscaling.
+ Fixes #402076.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Prefer apev1/2 and id3v1 (at end of file) over musepack.
+ tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
+ (eos_buffer_probe):
+ Add a test that ensures we set DELTA_UNIT on all non-header,
+ non-video buffers, if we have a video stream.
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
+ (gst_ogg_mux_process_best_pad):
+ Move setting delta_pad to earlier, where we inspect all pads, so
+ that leading audio pages don't get DELTA_UNIT unset if they come
+ before the first DELTA_UNIT from video pages. Fixes the newly-added
+ test. Fixes #385527.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream):
- Signal no-more-pads (so it works in playbin).
+ tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
+ Original commit message from CVS:
+ * tests/check/pipelines/streamheader.c: (streamheader_suite):
+ Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
+ fails on the p5-ppc64 build bot and the failure looks like it is due
+ to the same issue as #348114, ie. a compiler bug.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstqueue2.c: Fix build on MacOSX.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_create_read):
+ Fix build on MacOSX.
+
+2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
+ Fix compilation on mingw. Fixes #446972.
+
+2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
+ Original commit message from CVS:
+ Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst/playback/gstqueue2.c: (update_buffering),
+ (gst_queue_locked_enqueue):
+ Fix a division by zero when the max percent is <= 0. Fixes #446572.
+ also update the buffering status when receiving events. Fixes #446551.
+
+2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
+
+ gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
+ Original commit message from CVS:
+ Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst/playback/gstqueue2.c: (gst_queue_peer_query),
+ (gst_queue_handle_src_query):
+ Wait for preroll before attempting to forward a duration query upstream.
+ Fixes #445505.
+
+2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Use G_GINT64_CONSTANT macro for int64 constant.
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgsttag.def:
+ Add new exported functions.
- * ext/musepack/gstmusepackreader.cpp:
- Workaround for older core.
+2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
+ The BOS page of the first Dirac video stream needs to come before
+ the BOS page of any Vorbis streams or other audio streams, just like
+ it is with Theora.
- * gst/ffmpegcolorspace/imgconvert.c: (yuv420p_to_yuv422):
- Actually test for odd width/height rather than testing whether
- a temporary variable that was 0 before we subtracted 1 is now
- not equal to zero (which it always is).
+2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Fix compilation.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_get_range):
+ Fix compilation.
+
+2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
+
+ gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
+ Original commit message from CVS:
+ Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst/playback/gstqueue2.c: (gst_queue_init),
+ (gst_queue_handle_sink_event), (gst_queue_chain),
+ (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
+ (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
+ (gst_queue_src_activate_pull):
+ Add pull based scheduling and fix some deadlocks. Fixes #444523.
+ Does not yet completely work because duration queries upstream won't
+ block yet.
+
+2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Some more fseeko checks.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/playback/gstqueue2.c: (gst_queue_create_read):
+ Some more fseeko checks.
+
+2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ configure.ac: check for large file support.
+ Original commit message from CVS:
+ * configure.ac:
+ check for large file support.
+
+2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
+
+ gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
+ Original commit message from CVS:
+ Based on a patch by Sven Arvidsson <sa at whiz dot se>:
+ * gst/subparse/gstsubparse.c: (parse_subrip),
+ (subviewer_unescape_newlines), (parse_subviewer),
+ (gst_sub_parse_data_format_autodetect),
+ (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
+ * gst/subparse/gstsubparse.h:
+ Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
+ * tests/check/elements/subparse.c: (GST_START_TEST),
+ (subparse_suite):
+ Add a unit test for both SubViewer formats.
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+ gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
+ Don't overflow intermediate values when seeking to large time values
+ in audiotestsrc.
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+ gst/playback/gstqueue2.c: Include stdio to define fseeko.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_have_data),
+ (gst_queue_create_read), (gst_queue_read_item_from_file),
+ (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
+ Include stdio to define fseeko.
-2004-11-11 Jan Schmidt <thaytan@mad.scientist.com>
+2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
- * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
- (gst_goom_dispose), (gst_goom_sinkconnect), (gst_goom_chain),
- (gst_goom_change_state), (plugin_init):
- Use the bytestream adapter so goom doesn't depend on the input
- buffer size.
- Add a debug category
+ sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
+ Original commit message from CVS:
+ Patch by: Edward Hervey <edward@fluendo.com>
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
+ (gst_v4lsrc_query):
+ Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsa.c: (gst_alsa_change_state):
- Only set hardware parameters *after* negotiation. Before
- negotiation, it will set ANY and that seems to cause crashes
- (see e.g. #151288, #153227).
+ gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
+ Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
+ our own implementation.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state):
+ Handle timestamp wraparound.
+
+2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c: (no_more_pads_full),
+ (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
+ (gst_uri_decode_bin_change_state):
+ Make sure we name srcpads uniquely even when using different internal
+ decodebins.
+ Signal no-more-pads when no more dynamic elements exist.
+ Remove pads on cleanup.
+
+2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
+
+ gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
+ Original commit message from CVS:
+ Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst/playback/gstqueue2.c: (gst_queue_class_init),
+ (gst_queue_init), (gst_queue_finalize),
+ (gst_queue_write_buffer_to_file), (gst_queue_have_data),
+ (gst_queue_create_read), (gst_queue_read_item_from_file),
+ (gst_queue_open_temp_location_file),
+ (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
+ (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
+ (gst_queue_is_empty), (gst_queue_is_filled),
+ (gst_queue_change_state), (gst_queue_set_temp_location),
+ (gst_queue_set_property):
+ Add support for filebased buffering. Fixes #441264.
+
+2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
+ (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
+ (caps_notify_group_cb), (gst_decode_group_new),
+ (gst_decode_group_free):
+ Add support for delayed caps fixation when autoplugging.
+ Optimize cases where a multiqueue is not needed/wanted, like right after
+ anything that is not a demuxer.
+
+2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
+ (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
+ consideratly speedup ogg chain detection by not trying to find a base
+ timestamp for skeleton streams.
+
+2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
+ (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_remove_flush),
+ (gst_multi_fd_sink_remove_client_link),
+ (gst_multi_fd_sink_handle_client_write),
+ (gst_multi_fd_sink_handle_clients):
+ * gst/tcp/gstmultifdsink.h:
+ Add support for remuve_flush.
+
+2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add draft design for forcing keyframes in encoders and implement in theoraenc.
+ Original commit message from CVS:
+ * docs/design/draft-keyframe-force.txt:
+ * ext/theora/theoraenc.c: (theora_enc_sink_event),
+ (theora_enc_chain):
+ Add draft design for forcing keyframes in encoders and implement in
+ theoraenc.
+
+2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
- This seems to be antique leftover. It needs to pass error
- checking.
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
- (gst_sdlvideosink_deinitsdl), (gst_sdlvideosink_initsdl),
- (gst_sdlvideosink_destroy), (gst_sdlvideosink_create),
- (gst_sdlvideosink_sinkconnect), (gst_sdlvideosink_chain):
- Fix GstXOverlay implementation (#151059).
+=== release 0.10.13 ===
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Don't assert (#157853).
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ * win32/vs6/grammar.dsp:
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstadder.dsp:
+ * win32/vs6/libgstaudio.dsp:
+ * win32/vs6/libgstaudioconvert.dsp:
+ * win32/vs6/libgstaudiorate.dsp:
+ * win32/vs6/libgstaudioresample.dsp:
+ * win32/vs6/libgstaudioscale.dsp:
+ * win32/vs6/libgstaudiotestsrc.dsp:
+ * win32/vs6/libgstcdda.dsp:
+ * win32/vs6/libgstdecodebin.dsp:
+ * win32/vs6/libgstdecodebin2.dsp:
+ * win32/vs6/libgstdirectsound.dsp:
+ * win32/vs6/libgstffmpegcolorspace.dsp:
+ * win32/vs6/libgstgdp.dsp:
+ * win32/vs6/libgstinterfaces.dsp:
+ * win32/vs6/libgstnetbuffer.dsp:
+ * win32/vs6/libgstogg.dsp:
+ * win32/vs6/libgstpbutils.dsp:
+ * win32/vs6/libgstplaybin.dsp:
+ * win32/vs6/libgstriff.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgstsinesrc.dsp:
+ * win32/vs6/libgstsubparse.dsp:
+ * win32/vs6/libgsttag.dsp:
+ * win32/vs6/libgsttheora.dsp:
+ * win32/vs6/libgsttypefindfunctions.dsp:
+ * win32/vs6/libgstutils.dsp:
+ * win32/vs6/libgstvideo.dsp:
+ * win32/vs6/libgstvideorate.dsp:
+ * win32/vs6/libgstvideoscale.dsp:
+ * win32/vs6/libgstvideotestsrc.dsp:
+ * win32/vs6/libgstvolume.dsp:
+ * win32/vs6/libgstvorbis.dsp:
+ Release 0.10.13 "What's going on?"
+ Original commit message from CVS:
+ Release 0.10.13 "What's going on?"
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
- Fix bytes/samples confustion.
- (gst_alsa_sink_mmap), (gst_alsa_sink_loop):
- Fix for underrun (#144389).
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
+
+ gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ In riff, the depth is stored in the size field but it just means that
+ the least significant bits are cleared. We can therefore just play
+ the sample as if it had a depth == width. Fixes: #440997
+ Patch by: Wim Taymans <wim@fluendo.com>
+ Patch by: Sebastian Dröge <slomo@circular-chaos.org>
+
+2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
+ Original commit message from CVS:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Define inline when needed on win32 builds. Fixes: #441295
+
+2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (queue_overrun),
+ (no_more_pads_full):
+ Stop buffering when the group is commited because the queues filled up.
+ Fixes #442024.
+
+2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
+ (gst_alsa_mixer_free), (gst_alsa_mixer_update),
+ (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
+ (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
+ (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
+ * ext/alsa/gstalsamixer.h:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_interface_supported),
+ (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
+ (gst_alsa_mixer_element_set_property),
+ (gst_alsa_mixer_element_get_property),
+ (gst_alsa_mixer_element_change_state):
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
+ * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
+ (gst_mixer_option_changed):
+ * gst-libs/gst/interfaces/mixer.h:
+ Revert commits towards #152864 made so far. We'll pick it up again
+ after the 0.10.13 release.
+
+2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ After an interrupt (PAUSED/flush) assume that the next sample should not
+ be aligned to the previous sample. Fixes #417992.
+
+2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Don't add channels and rate fields to the template caps for
+ audio/x-dts, as wavparse might not always be able to set them,
+ which would then lead to 'caps are not a real subset of the
+ template caps' warnings.
-2004-11-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
+ Handle unknown or invalid pads without crashing, as might occur if
+ a media file like an mp3 is specified as a subtitle file.
+ Fixes: #410039
+
+2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
+ (setup_sinks):
+ Block the subtitle bin output queue before ghosting it and linking,
+ then unblock after. This avoids spurious not-linked errors caused
+ by the queue starting up (because it gets linked when it is ghosted).
+ Fixes: #350299
+
+2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
+ Original commit message from CVS:
+ * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
+ Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
+ file. Avoids flukes where the input gets typefound to some valid but
+ useless type.
+
+2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
+ (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
+ Add unit test for gnomevfssink seeking and position reporting for
+ file:// URIs.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- Disable halfway-seek for pending release (since it needs a new
- core release).
+2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/gnomevfs/gstgnomevfssink.*: see #412648.
+ Original commit message from CVS:
+ Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
+ (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
+ (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Fix position reporting, especially after a seek (from upstream),
+ see #412648.
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- add autoprobe-fps property so we can separate autoprobing parts
+2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ Repair umlaut.
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/v4lsrc_calls.c:
- initialise fourcc to catch unset fourcc's, and debug
-
-2004-11-09 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (close_pad_link), (try_to_link_1):
- * gst/playback/gstplaybin.c: (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_property),
- (remove_sinks), (setup_sinks), (gst_play_bin_change_state),
- (gst_play_bin_get_event_masks), (gst_play_bin_send_event),
- (gst_play_bin_get_formats), (gst_play_bin_convert),
- (gst_play_bin_get_query_types), (gst_play_bin_query):
- Cleanups and some more documentation.
-
-2004-11-09 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init),
- (gst_cacasink_init), (gst_cacasink_chain):
- * ext/libcaca/gstcacasink.h:
- Cacasink inherits from VideoSink, so let that store the clock.
-
-2004-11-09 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/README:
- * gst/playback/gstplaybasebin.c: (group_destroy), (group_is_muted),
- (add_stream), (unknown_type), (add_element_stream), (no_more_pads),
- (probe_triggered), (preroll_unlinked), (new_decoded_pad),
- (gst_play_base_bin_change_state), (gst_play_base_bin_found_tag):
- * gst/playback/gstplaybin.c: (gen_vis_element), (remove_sinks),
- (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute),
- (gst_stream_info_is_mute), (gst_stream_info_set_property):
- * gst/playback/gststreaminfo.h:
- Updated README.
- Only switch groups if all streams have muted (EOSed).
- Send Tags in sync with the stream playback instead of in
- the playback/preroll phase.
- Some cleanups, free the fakesrc elements.
-
-2004-11-09 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
- buffer-frames property was missing
- * ext/arts/gst_arts.c:
- rate missing from sinkcaps
- * ext/audiofile/gstafparse.c:
- * ext/audiofile/gstafsink.c:
- * ext/audiofile/gstafsrc.c:
- * ext/swfdec/gstswfdec.c:
- int audio doesn't know buffer-frames
- * ext/cdparanoia/gstcdparanoia.c:
- int audio doesn't know chunksize either
- * ext/nas/nassink.c:
- it's endianness, not endianess
- * gst-libs/gst/audio/audio.h:
- make float standard pad template caps really describe float
- * gst/law/mulaw.c: (linear_factory):
- signed only, please
- * gst/mpegstream/gstdvddemux.c:
- widths of 20 are not valid
+2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-11-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Specify the full valid range for MP3 samplerates. Fixes a regression
+ caused by extra header checks since the last release.
- Submitted by: Luca Ferretti <elle.uca@infinito.it>
+2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
- * po/LINGUAS:
- * po/it.po:
- Add Italian
+ sys/: Fix a locking-order bug I introduced with my changes the other day.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
+ Fix a locking-order bug I introduced with my changes the other day.
+ Patch by Mike Smith.
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (close_pad_link), (try_to_link_1):
- * gst/playback/gstplaybasebin.c: (probe_triggered),
- (gst_play_base_bin_change_state):
- Updated README, added more comments for fixmes etc..
+ ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_data_packet):
+ Don't look inside 0-length packets (which indicate duplicated
+ frames)
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Small cleanups.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_read_sector):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Small cleanups.
+ * ext/theora/theoradec.c: (theora_dec_sink_event):
+ Fix typo.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Add some FIXME
+ * gst/playback/gstdecodebin.c: (queue_underrun_cb):
+ And some debug info when a FIXME path is hit.
+
+2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_class_init),
+ (gst_base_rtp_audio_payload_init),
+ (gst_base_rtp_audio_payload_finalize),
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer),
+ (gst_base_rtp_payload_audio_handle_event):
+ Some cleanups, remove minptime property as it is now in the parent
+ class.
+ Override parent class event function.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init), (gst_basertppayload_init),
+ (gst_basertppayload_event), (gst_basertppayload_set_property),
+ (gst_basertppayload_get_property):
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Add min-ptime property.
+ Add handle-event vmethod. Fixes #415001.
+
+2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- We can remove this hack now.
+ * gst-plugins-base.spec.in:
+ update spec
+ Original commit message from CVS:
+ update spec
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
- (gst_videomixer_fill_checker), (gst_videomixer_fill_color),
- (gst_videomixer_blend_buffers), (gst_videomixer_loop):
- Only mix AYUV for maximum quality.
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_change_state):
+ Fix typo in comment.
+ * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
+ free_dynamics, pad_probe, close_pad_link, try_to_link_1,
+ get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
+ close_link):
+ * gst/playback/gstplaybin.c (gst_play_bin_set_property,
+ gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
+ Remove trailing whitespaces in comments.
+ * gst/volume/Makefile.am:
+ Fix tabs.
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
- * ext/ogg/gstoggdemux.c: (get_relative), (gst_ogg_demux_src_query),
- (gst_ogg_demux_push), (gst_ogg_pad_push):
- Let's act as if we're synchronized now! :).
- * ext/theora/theoradec.c: (theora_dec_chain):
- Add some debug.
+ * ChangeLog:
+ * gst-libs/gst/interfaces/mixer.h:
+ gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
+ Original commit message from CVS:
+ patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
+ set_option, get_option, _gst_reserved):
+ Revert reordering functions (keep ABI).
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: When we create our own window, indicate that we handle the
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
+ (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
+ (gst_ximagesink_show_frame):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
+ (gst_xvimagesink_show_frame):
+ When we create our own window, indicate that we handle the
+ WM_DELETE client message from the window manager, so that it won't
+ kill our window (and our app) along with it. Handle ClientMessage,
+ post an error on the bus, and close the window. Further buffers
+ arriving will result in a FlowError because the window has been
+ destroyed.
+ Fixes: #393975
+ Clean up the X event handling loop and make them the same for
+ both xvimagesink and ximagesink while I'm at it.
+
+2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
+ Make decodebin2 autoplug depayloaders too.
+ * gst/playback/gsturidecodebin.c: (source_new_pad):
+ Set the newly created decoder in a usable state when autoplugging a
+ dynamic source such as RTSP.
+
+2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c: (cb_probe):
+ Ignore video-codec tag for audio streams and ignore audio-codec tags
+ for video streams. Should make codec name collection a bit more
+ robust against sloppy demuxers that send tag events containing both
+ tags down each pad.
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_set_property), (gst_alpha_sink_link),
- (gst_alpha_set_ayuv), (gst_alpha_set_i420),
- (gst_alpha_chroma_key_ayuv), (gst_alpha_chroma_key_i420),
- (gst_alpha_init_params), (gst_alpha_chain):
- Implement alpha functions for AYUV too, this increases
- accuracy quite a bit.
+2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (update_rates):
+ Tweak the buffering thresholds a little.
+ Update the buffer size with the previously calculate rate instead of
+ only when we calculate a new rate so that we get smoother buffering
+ updates.
+ * gst/playback/Makefile.am:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
+ (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
+ (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
+ (gst_uri_decode_bin_get_property), (unknown_type),
+ (add_element_stream), (no_more_pads_full), (no_more_pads),
+ (source_no_more_pads), (new_decoded_pad), (array_has_value),
+ (gen_source_element), (has_all_raw_caps), (analyse_source),
+ (remove_decoders), (make_decoder), (remove_source),
+ (source_new_pad), (setup_source), (decoder_query_init),
+ (decoder_query_duration_fold), (decoder_query_duration_done),
+ (decoder_query_position_fold), (decoder_query_position_done),
+ (decoder_query_latency_fold), (decoder_query_latency_done),
+ (decoder_query_seeking_fold), (decoder_query_seeking_done),
+ (decoder_query_generic_fold), (gst_uri_decode_bin_query),
+ (gst_uri_decode_bin_change_state), (plugin_init):
+ New element that intergrates a source, optional buffering element and
+ decodebin.
+
+2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
+ Original commit message from CVS:
+ * configure.ac:
+ Bump libtheora requirement to 1.0alpha5 for the pixformat check
+ (also has a .pc file, so we don't need the fallback check any
+ longer). Fixes #438840.
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstqueue2.c: fix build.
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c: (gst_queue_get_type),
+ (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
+ (apply_segment), (apply_buffer), (update_buffering),
+ (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
+ (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
+ (gst_queue_handle_sink_event), (gst_queue_is_filled),
+ (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
+ (plugin_init):
+ fix build.
+
+2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstqueue2.c: (gst_queue_get_type),
+ (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
+ (gst_queue_getcaps), (gst_queue_bufferalloc),
+ (gst_queue_acceptcaps), (update_time_level), (apply_segment),
+ (apply_buffer), (update_buffering), (reset_rate_timer),
+ (update_rates), (gst_queue_locked_flush),
+ (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
+ (gst_queue_handle_sink_event), (gst_queue_is_empty),
+ (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
+ (gst_queue_loop), (gst_queue_handle_src_event),
+ (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
+ (gst_queue_src_activate_push), (gst_queue_change_state),
+ (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
+ On our way to playbin2 this is the new network queue that does buffering
+ all by itself using high and low watermarks. It can also measure up and
+ downstream bandwidth to optimally size the queue.
+
+2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
+ Use the segment->last_stop value to calculate the next timestamp to
+ generate after a seek; not the segment->start value.
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (shrink12), (img_get_alpha_info), (deinterlace_line),
- (deinterlace_line_inplace):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Added AYUV colorspace and handle RGBA a bit more respectful.
+2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
+ Original commit message from CVS:
+ * docs/Makefile.am: Install docs even when --disable-gtk-doc
+ is disabled. This matches the behavior of gtk+. Fixes #349099.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Actually always send a discont (cornercase when resending the
- same serial-tagged chain twice).
+2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-08 Julien MOUTTE <julien@moutte.net>
+ ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
+ Some more chained streaming ogg timestamp fixes.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_finalize):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_finalize): Some more cleanups, leaks fixed and checks.
+2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ ext/ogg/gstoggdemux.c: Add some FIXMEs.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_handle_page):
+ Add some FIXMEs.
+ Fix chain start/stop segment handling based on patch by
+ <ahalda at cs dot mcgill dot ca> see #320984.
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
- Don't segfault on NULL data.
+2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ configure.ac: We don't require a C++ compiler. So don't require one.
+ Original commit message from CVS:
+ * configure.ac:
+ We don't require a C++ compiler. So don't require one.
- * gst/playback/gstdecodebin.c: (unlinked):
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (group_create), (get_active_group),
- (get_building_group), (group_destroy), (group_commit),
- (queue_overrun), (remove_groups), (add_stream), (unknown_type),
- (add_element_stream), (no_more_pads), (probe_triggered),
- (preroll_unlinked), (new_decoded_pad), (removed_decoded_pad),
- (state_change), (setup_source), (gst_play_base_bin_get_property),
- (gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
- (gst_play_base_bin_link_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (remove_sinks), (setup_sinks), (gst_play_bin_change_state):
- Add support for chained ogg files. Prepare for playlist
- support. This patch introduces the concept of pad groups, which
- together compose one playable media file.
+2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-07 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ * ext/alsa/gstalsamixer.c:
+ ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
+ gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
+ gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
+ gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
+ gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
+ gst_alsa_mixer_update_track):
+ Apply some of the cleanup Tim suggested in #152864 afterwards.
- * testsuite/gst-lint: Check for pad templates that aren't statically
- scoped.
+2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
+ Original commit message from CVS:
+ patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
+ _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
+ gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
+ gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
+ gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
+ gst_alsa_mixer_handle_source_callback,
+ gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
+ gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
+ gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
+ gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
+ gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
+ gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
+ * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
+ * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
+ gst_alsa_mixer_element_interface_supported,
+ gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
+ gst_alsa_mixer_element_set_property,
+ gst_alsa_mixer_element_get_property,
+ gst_alsa_mixer_element_change_state):
+ * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
+ * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
+ gst_mixer_option_changed):
+ * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
+ volume_changed, option_changed, _gst_reserved):
+ Implement notification for alsamixer. Fixes #152864
- * configure.ac:
- * ext/Makefile.am:
- * ext/musepack/Makefile.am:
- * ext/musepack/gstmusepackdec.cpp:
- * ext/musepack/gstmusepackdec.h:
- * ext/musepack/gstmusepackreader.cpp:
- * ext/musepack/gstmusepackreader.h:
- Add musepack decoder.
- * ext/faad/gstfaad.c: (gst_faad_base_init):
- Make pad templates static.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (plugin_init):
- Add musepack typefinder, make mp3 typefinding work halfway stream,
- which doesn't actually work yet because id3demux doesn't implement
- _get_length().
+2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ Add support for video/x-raw-bayer.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_loop):
- Fix interrupt event handling (#144436).
+2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Add some sanity checking for the XVImage size returned by X.
+ Related to #377400.
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_typefind):
- Hide unused glory.
+2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Parse and use additional caps fields as described in updated
+ application/x-rtp caps spec.
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/vorbis/vorbisenc.c: (raw_caps_factory):
- Fix weird caps (#157548).
+ ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_collect_chain_info):
+ If there is a stream in a chain without any data packets, ignore the
+ stream in the total length calculations. Might be related to #436820.
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
+ (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
+ (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
+ (mpeg_video_type_find), (mpeg_video_stream_type_find),
+ (plugin_init):
+ Consolidate and re-work our mpeg system stream detection to probe
+ more packets and produce a higher confidence result. Fixes a
+ regression caused by lowering the typefind probability last year
+ - related to bug #397810. Remove the redundant MPEG-1 specific
+ typefind function, as the new one detects both MPEG-1 & MPEG-2
+ happily.
+ Also cleanup the MPEG elementary and MPEG-TS detection functions a
+ little.
+ Tested against my media test directory, with some improvements and
+ no regressions.
+
+2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
+ (queue_out_of_data):
+ Connect to the new queue "pushing" signal instead of the broken
+ "running" one.
+
+2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer):
+ Move variable declaration before the first instruction.
+ * gst/videotestsrc/videotestsrc.c:
+ Define M_PI if it's not defined yet.
+ * win32/common/libgstrtp.def:
+ Add new exported functions.
- * gst/rtp/gstrtpgsmparse.c: (gst_rtpgsm_caps_nego):
- Add missing NULL terminator (#157543).
+2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
-2004-11-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_type_packet):
+ gst_pad_push_event() does not return a GstFlowReturn!
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- ports can go up to 65535. Move common defines to gsttcp.h
+2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ tests/examples/seek/: Some small cosmetic changes.
+ Original commit message from CVS:
+ * tests/examples/seek/scrubby.c: (stop_cb), (main):
+ * tests/examples/seek/seek.c: (do_seek):
+ Some small cosmetic changes.
- * gst/videotestsrc/videotestsrc.c: (paint_setup_Y41B),
- (paint_hline_Y41B), (paint_setup_Y42B), (paint_hline_Y42B):
- Added two more colorspaces.
+2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
+ gst_adder_change_state):
+ * gst/adder/gstadder.h (bps, offset, collect_event, segment,
+ segment_pending, segment_position, segment_rate):
+ Handle playback-rate on adder.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (yuv422p_to_yuv422), (yuv420p_to_yuv422), (shrink12),
- (img_convert), (deinterlace_line), (deinterlace_line_inplace):
- More stride fixes.
+2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_reset),
+ (theora_dec_sink_event), (theora_handle_comment_packet),
+ (theora_handle_type_packet), (theora_dec_change_state):
+ Don't push events (newsegment, tags) before initialising the
+ decoder.
+ This is neccesary for seeking to work correctly in gnonlin.
- * gst/alpha/gstalpha.c: (gst_alpha_set_property), (gst_alpha_add),
- (gst_alpha_chroma_key), (gst_alpha_init_params), (gst_alpha_chain):
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_i420),
- (gst_videomixer_fill_checker), (gst_videomixer_blend_buffers),
- (gst_videomixer_loop):
- More stride fixes.
+2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-05 Benjamin Otte <otte@gnome.org>
+ gst/: gst/audiotestsrc/gstaudiotestsrc.c
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst/adder/gstadder.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c
+ (gst_audio_test_src_create_white_noise):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
+ VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
+ volume_sink_template, volume_src_template, gst_volume_init,
+ volume_process_double, volume_process_int16,
+ volume_process_int16_clamp):
+ Doc fixes and formatting.
- * ext/mad/gstmad.c: (gst_mad_chain):
- don't overflow data buffer. Flush not needed sync data when syncing
- failed.
+2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-04 Wim Taymans <wim@fluendo.com>
+ tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
+ Minimal check for volume's GstController usability; also another
+ test for #422295.
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_class_init), (gst_alpha_init), (gst_alpha_set_property),
- (gst_alpha_get_property), (gst_alpha_add), (gst_alpha_chroma_key),
- (gst_alpha_init_params), (gst_alpha_chain),
- (gst_alpha_change_state):
- Updated the chroma keying algorithm with something more
- sophisticated.
+2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_add_track):
+ Fix it so that it (a) makes sense and (b) doesn't break
+ everything cdda-related including the unit test.
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_i420),
- (gst_videomixer_fill_checker), (gst_videomixer_fill_color),
- (gst_videomixer_blend_buffers), (gst_videomixer_loop):
- Fix stride issues. Does not completely work for odd
- heights.
+2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_add_track):
+ Fix build when disabling asserts.
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_chroma_key), (gst_alpha_chain):
- Fix stride issues. Does not completely work for odd
- heights.
+2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-03 Christophe Fergeau <teuf@gnome.org>
+ sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
+ When XShm is not available, we might get row strides that are not
+ rounded up to multiples of four; this is bad, because virtually
+ every RGB-processing element in GStreamer assumes rowstrides are
+ rounded up to multiples of four, so let's allocate at least enough
+ memory to avoid crashes in this case. The image will still be
+ displayed distorted though if this happens, so that still needs
+ fixing (maybe by allocating a bigger image with an 'even' width
+ and then clipping it appropriately when rendering - something for
+ Xlib aficionados in any case).
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- * gst/videoscale/videoscale.c: (videoscale_find_by_structure):
- leak fixes
+2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ If a buffer doesn't have a timestamp, assume it's contiguous with
+ the previous buffer, and synthesise timestamps appropriately.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (avpicture_alloc):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Use correct _fill function to get correct strides.
-
-2004-11-02 David Schleef <ds@schleef.org>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
- (qtdemux_parse_udta), (qtdemux_tag_add), (gst_qtdemux_handle_esds):
- Change all g_print()s to debugging. Add a bunch of consistency
- checks.
-
-2004-11-02 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (try_to_link_1), (get_our_ghost_pad), (remove_element_chain),
- (unlinked), (no_more_pads), (close_link):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (unknown_type), (add_element_stream), (new_decoded_pad),
- (removed_decoded_pad), (setup_source):
- * gst/playback/gststreaminfo.c: (gst_stream_info_get_type),
- (gst_stream_info_class_init), (gst_stream_info_init),
- (gst_stream_info_new), (gst_stream_info_dispose),
- (stream_info_mute_pad), (gst_stream_info_set_property),
- (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Fix playback of multiple files.
- a slightly different approach to handling dynamic pad removals.
- This one only looks at pads that we have linked.
-
-2004-11-01 Christophe Fergeau <teuf@gnome.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_finalize): fix an "invalid
- free" warning from libc.
-
-2004-11-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (get_unconnected_element), (remove_starting_from), (pad_removed),
- (close_link):
- Implement support for dynamic pad changing. We listen to "live"
- pad removals (i.e. while playing) and re-setup autoplugging
- after that. Playbasebin/playbin need some more work for this
- to finally work, but decodebin supports (and replugs) chained
- ogg now.
-
-2004-11-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
- (gst_alsa_finalize):
- * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
- (gst_cdaudio_finalize):
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_finalize):
- * ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
- * ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
- (dvdreadsrc_finalize):
- * ext/flac/gstflacdec.c: (gst_flacdec_class_init),
- (gst_flacdec_finalize):
- * ext/flac/gstflacenc.c: (gst_flacenc_class_init),
- (gst_flacenc_finalize):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
- (gst_gnomevfssink_finalize):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
- (gst_gnomevfssrc_finalize):
- * ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
- (gst_fameenc_finalize):
- * ext/nas/nassink.c: (gst_nassink_class_init),
- (gst_nassink_finalize):
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
- (gst_sdlvideosink_class_init):
- * ext/sndfile/gstsf.c: (gst_sf_dispose):
- * gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
- * gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
- * gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
- * gst-libs/gst/xwindowlistener/xwindowlistener.c:
- (gst_x_window_listener_dispose):
- * gst/audioscale/gstaudioscale.c:
- * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
- (play_on_demand_finalize):
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
- * sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
- (cdplayer_finalize):
- * sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
- (gst_glimagesink_class_init):
- * sys/oss/gstosselement.c: (gst_osselement_class_init),
- (gst_osselement_finalize):
- * sys/oss/gstosssink.c: (gst_osssink_dispose):
- * sys/oss/gstosssrc.c: (gst_osssrc_dispose):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
- Fixes a bunch of problems with finalize and dispose functions,
- either assumptions that dispose is only called once, or not calling
- the parent class dispose/finalize function
-
-2004-11-01 Stefan Kost <ensonic@users.sf.net>
-
- * ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_link):
- added two api precondition guards
- use g_strdup with getenv to fix crash when using ENVVAR
-
-2004-11-01 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/esd/esdsink.c: (gst_esdsink_class_init),
- (gst_esdsink_finalize):
- Use a finalize function, not dispose, and more importantly,
- call the parent class finalize function too
-
-2004-11-01 Johan Dahlin <johan@gnome.org>
+2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/ogg/gstoggdemux.c:
- * gst/tags/gstvorbistag.c:
- Plug leaks.
+ tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
+ Original commit message from CVS:
+ * tests/check/elements/videorate.c: (GST_START_TEST):
+ Set buffer timestamp to a valid value in order to test the buffer
+ really does stay in videorate.
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- lotsa memleaks today. But they're all small...
+ gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
+ There is no sensible way to handle incoming buffers which don't have a
+ valid timestamp. We therefore discard them and wait for the next one.
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- another memleak crushed
+ gst/playback/: Better error message for text files.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
+ * gst/playback/gstdecodebin2.c: (plugin_init):
+ Better error message for text files.
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
+ Fix offset bug in generation RR packets.
+
+2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
+ Original commit message from CVS:
+ 2007-04-27 Julien MOUTTE <julien@moutte.net>
+ * ext/theora/theoradec.c: (_theora_granule_time),
+ (theora_dec_push_forward), (theora_handle_data_packet),
+ (theora_dec_decode_buffer): Calculate buffer duration correctly
+ to generate a perfect stream (#433888).
+ * gst/audioresample/gstaudioresample.c:
+ (audioresample_check_discont): Glib provides ABS.
+
+2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
+ (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
+ (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
+ (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
+ (gst_rtcp_packet_bye_set_reason):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Fix RB block parsing and writing.
+ Add support for constructing BYE packets.
+
+2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ When posting a warning message because samples were dropped, post something more intelligible than he default error m...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
+ (gst_base_audio_src_create):
+ * po/POTFILES.in:
+ When posting a warning message because samples were dropped, post
+ something more intelligible than he default error message for clock
+ errors which is just confusing in this context (#432984).
- * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- fix memleak
+2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
+ (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
+ (read_packet_header), (gst_rtcp_packet_move_to_next),
+ (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
+ (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
+ (gst_rtcp_packet_sdes_get_item_count),
+ (gst_rtcp_packet_sdes_first_item),
+ (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
+ (gst_rtcp_packet_sdes_first_entry),
+ (gst_rtcp_packet_sdes_next_entry),
+ (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
+ (gst_rtcp_packet_sdes_add_entry):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Implement code to write SR, RR and SDES packets.
+
+2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
+
+ sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
+ Original commit message from CVS:
+ Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
+ * sys/ximage/ximagesink.c:
+ Fix build if XShm is not available (#432362).
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Hack to prevent crash when going to READY inside signal handler
- while this function is active.
+ gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
+ Initalize the AudioConvertCtx with zeroes, otherwise it will contain
+ pointers to random memory which are passed to g_free() when
+ audio_convert_prepare_context() is called the first time.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/common.h:
- * gst/ffmpegcolorspace/dsputil.c: (dsputil_static_init):
- * gst/ffmpegcolorspace/dsputil.h:
- * gst/ffmpegcolorspace/gstffmpeg.c: (plugin_init):
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_get_palette), (gst_ffmpeg_set_palette),
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps),
- (gst_ffmpegcsp_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt),
- (gst_ffmpeg_caps_to_pixfmt), (gst_ffmpegcsp_caps_with_codectype),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info), (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_pad_link), (gst_ffmpegcsp_get_type),
- (gst_ffmpegcsp_base_init), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_chain),
- (gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
- (gst_ffmpegcsp_get_property), (gst_ffmpegcolorspace_register):
- * gst/ffmpegcolorspace/imgconvert.c:
- (avcodec_get_chroma_sub_sample), (avcodec_get_pix_fmt_name),
- (avcodec_get_pix_fmt), (avpicture_fill), (avpicture_layout),
- (avpicture_get_size), (avcodec_get_pix_fmt_loss),
- (avg_bits_per_pixel), (avcodec_find_best_pix_fmt1),
- (avcodec_find_best_pix_fmt), (img_copy_plane), (img_copy),
- (yuv422_to_yuv420p), (uyvy422_to_yuv420p), (uyvy422_to_yuv422p),
- (yuv422_to_yuv422p), (yuv422p_to_yuv422), (yuv422p_to_uyvy422),
- (uyvy411_to_yuv411p), (yuv420p_to_yuv422), (C_JPEG_TO_CCIR),
- (img_convert_init), (img_apply_table), (shrink41), (shrink21),
- (shrink12), (shrink22), (shrink44), (grow21_line), (grow41_line),
- (grow21), (grow22), (grow41), (grow44), (conv411),
- (gif_clut_index), (build_rgb_palette), (bitcopy_n), (mono_to_gray),
- (monowhite_to_gray), (monoblack_to_gray), (gray_to_mono),
- (gray_to_monowhite), (gray_to_monoblack), (avpicture_alloc),
- (avpicture_free), (is_yuv_planar), (img_convert),
- (get_alpha_info_pal8), (img_get_alpha_info), (deinterlace_line),
- (deinterlace_line_inplace), (deinterlace_bottom_field),
- (deinterlace_bottom_field_inplace), (avpicture_deinterlace):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- * gst/ffmpegcolorspace/mem.c: (av_malloc), (av_realloc), (av_free):
- * gst/ffmpegcolorspace/mmx.h:
- * gst/ffmpegcolorspace/utils.c: (av_mallocz), (av_strdup),
- (av_fast_realloc), (av_mallocz_static), (av_free_static),
- (av_freep), (avcodec_get_context_defaults),
- (avcodec_alloc_context), (avcodec_init):
- Sync back from gst-ffmpeg. Deprecates ffcolorspace. Adds palette
- handling plus update from ffmpeg CVS. Large clean-up.
+ gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
+ Original commit message from CVS:
+ Patch by: Dan Williams <dcbw redhat com>
+ * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
+ Don't leak incoming buffer if gst_pad_push() returns a
+ non-OK flow. Fixes #432755.
+ * tests/check/elements/videorate.c: (GST_START_TEST),
+ (videorate_suite):
+ Unit test for the above by Yours Truly.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/Makefile.am:
- We need the marshallers for decodebin, too.
+ gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
+ (gst_adder_sink_event), (gst_adder_collected):
+ Fix non-flushing segmented seeks, Fixes #340060 for me
-2004-10-30 David Schleef <ds@schleef.org>
+2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find): Make
- quicktime typefinding work with 64-bit offsets.
+ * ChangeLog:
+ ChangeLog surgery: add API keyword
+ Original commit message from CVS:
+ ChangeLog surgery: add API keyword
+
+2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
+ Original commit message from CVS:
+ Patch by: Olivier Crete <tester at tester ca>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_class_init),
+ (gst_base_rtp_audio_payload_init),
+ (gst_base_rtp_audio_payload_dispose):
+ Chain up to parent class in dispose function; get rid of
+ unnecessary 'diposed' flag in private structure (#415001).
-2004-10-30 Jan Schmidt <thaytan@mad.scientist.com>
+2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
- Set EOS on the element when processing an EOS event.
- * ext/speex/gstspeexdec.h:
- * ext/speex/gstspeexenc.h:
- Only keep a const ptr to the mode
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data),
- (gst_riff_create_audio_template_caps):
- Allow WMAV3, with up to 6 channels.
- * gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
- Don't call gst_pad_set_event_function on a sink pad.
- * gst/mpegstream/gstdvddemux.c:
- (gst_dvd_demux_get_subpicture_stream),
- (gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
- Copy the explicit caps that were set across to the cur_* pads,
- instead of trying to use a possibly non-existent negotiated caps.
- Reset the type of subpicture pads to UNKNOWN after calling
- init_stream, so that the caps get set.
+ Some minor docs fixes and additions; also add missing 'Since' bits.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_class_init):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ Some minor docs fixes and additions; also add missing 'Since' bits.
-2004-10-29 Martin Pitt <martin.pitt@canonical.com>
+2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
+ Original commit message from CVS:
+ Patch by: Zeeshan Ali <zeenix gmail com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer),
+ (gst_base_rtp_audio_payload_push):
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ The recently-added gst_base_rtp_audio_payload_push() should take an
+ object of type GstBaseRTPAudioPayload as first argument (#431672).
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
- Don't touch buffer if it is of size 0 (fixes #151064).
+2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ Make more functions static, just because we can.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push), (gst_ogg_pad_push):
- Synchronized discont handling.
+2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c:
+ Add unit test for audioresample shutdown crasher (#420106).
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_push):
- Make seeking sort-of exact again (fixes #156387).
+2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/subparse/: Use GST_DISABLE_XML here
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ Use GST_DISABLE_XML here
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_buffer_alloc),
+ (gst_xvimagesink_navigation_send_event):
+ * sys/xvimage/xvimagesink.h:
+ Include stdlib.h when using atoi.
+ * tests/check/elements/playbin.c: (playbin_suite):
+ Use GST_DISABLE_REGISTRY here
- * gst/playback/gstplaybasebin.c: (unknown_type),
- (add_element_stream), (new_decoded_pad),
- (gst_play_base_bin_change_state):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Make caps explicitely available. Makes testing for unsupported
- types possible. Improves error reporting.
+2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
+ Original commit message from CVS:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
+ (theora_enc_sink_event), (theora_enc_change_state):
+ Track initialisation state; don't try to use encoder state if we're
+ not initialised (it'll segfault).
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format):
- Really don't touch read-only buffers (#156563).
+2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-29 Sebastien Cote <sc5@hermes.usherb.ca>
+ tests/check/pipelines/.cvsignore: Fix build.
+ Original commit message from CVS:
+ * tests/check/pipelines/.cvsignore:
+ Fix build.
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
+ Original commit message from CVS:
+ * gst/app/Makefile.am:
+ Fix CFLAGS and hopefully #430594.
+
+2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Allow random depths between 1 and 32 instead of only multiplies of 8.
+
+2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Set the maximum number of channels for PCM and float in the correct
+ place to have it also used when creating the template caps.
+
+2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Correctly support 4, 6 and 8 channels with normal PCM and float
+ wav files.
+ Fix the depth and signedness calculation in extensible wav files and
+ also handle 1, 2, 4, 6, 8 channels here when a file without channel
+ mask is found.
+ Add support for float, alaw and mulaw in extensible wav files.
+ This allows correct playback of all but 5 files from
+ http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
+ (gst_riff_create_audio_template_caps):
+ Add voxware and float formats to the template caps.
+
+2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
+
+ ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
+ Original commit message from CVS:
+ Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
+ Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
+ Use the correct format strings for integer formats.
+
+2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak (#155223).
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
-2004-10-29 Wim Taymans <wim@fluendo.com>
+2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/tcp/.cvsignore:
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multifdsink_class_init), (gst_multifdsink_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_remove_client_link), (is_sync_frame),
- (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Added burst on connect sync_method, deprecated sync_clients,
- streamlined the sync code some more.
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybasebin.c: (thread_error), (setup_source),
- (gst_play_base_bin_change_state):
- Improve error reporting.
+ ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
+ Original commit message from CVS:
+ * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
+ Don't use pad_alloc_buffer_and_set_caps to create a small header
+ packet, or, worse, to create a big temporary video buffer using the
+ src pad.
-2004-10-28 Wim Taymans <wim@fluendo.com>
+2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/tcp/Makefile.am:
- * gst/tcp/fdsetstress.c: (mess_some_more), (run_test), (main):
- * gst/tcp/gstfdset.c: (nearest_pow), (resize), (ensure_size),
- (gst_fdset_new), (gst_fdset_free), (gst_fdset_set_mode),
- (gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- Added more locks around fdset structures. Fixed/reworked
- the poll array resizing code.
- Added stress test for fdset.
-
-2004-10-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
- fix build
-
-2004-10-28 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
- fix link function to always query channels and query width for
- floats
- * configure.ac:
- add equalizer dir
- * gst/equalizer/Makefile.am:
- * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
- (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
- (gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
- (arg_to_scale), (setup_filter),
- (gst_iir_equalizer_compute_frequencies),
- (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
- (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
- (plugin_init):
- add an equalizer
-
-2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Submitted by: Kjartan Maraas <kmaraas@broadpark.no>
+ gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
+ * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
+ GST_START_TEST, buffer_probe_cb, GST_START_TEST):
+ Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
- * po/LINGUAS:
- * po/nb.po:
- Added Norwegian Bokmaal translation
+2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/tcp/gstmultifdsink.c:
+ add debug
+ Original commit message from CVS:
+ add debug
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- Don't break on options (fixes #156488).
+2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ * tests/check/pipelines/streamheader.c:
+ tests/check/pipelines/streamheader.c (tag_event_probe_cb,
+ Original commit message from CVS:
+ * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
+ GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
+ streamheader_suite):
+ Add another test set up for failure
- * configure.ac:
- * ext/cdaudio/Makefile.am:
- * sys/Makefile.am:
- fix build on older automake
+2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-26 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggmux.c:
+ * gst/gdp/gstgdpdepay.c:
+ debug changes
+ Original commit message from CVS:
+ debug changes
- * ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_push), (gst_dvdec_loop):
- Allow a little margin when negotiating the framerate.
+2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-26 Stefan Kost <ensonic@users.sf.net>
+ tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
+ GST_START_TEST, streamheader_suite, main):
+ Add a test for the streamheader bug Wim fixed.
- * gst/level/gstlevel.c:
- synchonised naming of pads and pad-templates
+2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-26 Wim Taymans <wim@fluendo.com>
+ ext/theora/theoradec.c: Fix misleading comment.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_sink_event):
+ Fix misleading comment.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_handle_event), (_find_chain_get_unknown_part),
- (_find_streams_check), (gst_ogg_demux_push):
- Fix EOS again. Needs to be done in a better way. We should not
- remove the pad if there is no new chained stream.
+2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-26 Iain <iaingnome@gmail.com>
+ gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ More sanity checks for the header fields.
- * ext/ogg/gstoggdemux.c (gst_ogg_pad_new): Free the tag list.
- * gst/audioscale/gstaudioscale.c (gst_audioscale_link): Free the copy
- of the caps.
- * gst/interleave/interleave.c (interleave_class_init): Hook up release
- pad.
- (interleave_release_pad): Remove the pad.
- * gst/level/gstlevel.c: Allow the level to take 1 or 2 channels.
- * sys/sunaudio/gstsunaudio.c (gst_sunaudio_setparams): Pay attention to
- the set device.
- * sys/xvimage/xvimagesink.c (gst_xvimagesink_get_xv_support): Free the
- attrs
- (gst_xvimagesink_xcontext_clear): Free the xcontext.
- (gst_xvimagesink_finalize): Free the par.
+2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
+ Try encodings from all environment variables, not just those in the
+ first environment variable that is set.
- * gst/avi/gstavimux.c: (gst_avimux_audsinkconnect),
- (gst_avimux_stop_file):
- First calculate the rate, and only then use it. Hdr.rate is a
- multiple and not a derivative of hdr.scale. Scale is not the
- same as blockalign but is solely related to rate.
+2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/videorate/gstvideorate.c: Add some debug.
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
+ (gst_video_rate_chain):
+ Add some debug.
+ * tests/check/elements/videorate.c: (GST_START_TEST),
+ (videorate_suite):
+ Added check for videorate changing caps handling. Closes #421834.
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_init),
- (gst_gnomevfssink_handle_event), (gst_gnomevfssink_chain):
- Implement seeking.
+2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
-2004-10-25 James Henstridge <james@jamesh.id.au>
+ ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ Use scale functions to avoid overflow when calculating duration of
+ vorbis buffers.
- Reviewed by: David Schleef <ds@schleef.org>
+2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * examples/gstplay/player.c: (got_stream_length), (main):
- * examples/seeking/cdplayer.c: (update_scale):
- * examples/seeking/seek.c: (format_value), (update_scale):
- * examples/seeking/spider_seek.c: (format_value), (update_scale),
- (stop_seek):
- Build fixes on AMD64.
+ API: add gst_tag_freeform_string_to_utf8() (#405072).
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
+ API: add gst_tag_freeform_string_to_utf8() (#405072).
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
+ Use gst_tag_freeform_string_to_utf8() here.
-2004-10-25 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- reviewed by: Ronald Bultje <rbultje at gnome dot org>
+ * gst/tcp/gstmultifdsink.c:
+ log tweaking
+ Original commit message from CVS:
+ log tweaking
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix for some v4l cards which hang in v4lsrc
+2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-25 Wim Taymans <wim@fluendo.com>
+ gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event):
+ Make sure we set the IN_CAPS flag correctly.
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
+ Get the IN_CAPS flag before we call functions that mess with the flags.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_remove),
- (gst_ogg_demux_push), (gst_ogg_chains_clear):
- Make sure to remove the pad when a new chain is
- encountered. Set some vars to NULL so we don't try
- to reference freed memory.
+2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-25 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ * gst/gdp/gstgdppay.c:
+ gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
+ gst_gdp_pay_chain, gst_gdp_pay_sink_event):
+ Only stamp buffers with offset/offset_end right before they get
+ pushed. This ensures offset continuity, which was not the case
+ before as shown by
+ gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
- * examples/seeking/Makefile.am:
- * examples/seeking/cdplayer.c: (update_scale):
- * examples/seeking/chained.c: (unlinked), (new_pad), (main):
- * examples/seeking/playbin.c: (make_playerbin_pipeline),
- (format_value), (update_scale), (iterate), (start_seek),
- (stop_seek), (print_media_info), (play_cb), (pause_cb), (stop_cb),
- (print_usage), (main):
- Added some more examples, update others.
+2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ adding debugging
+ Original commit message from CVS:
+ adding debugging
- * ext/flac/gstflacdec.c: (gst_flacdec_update_metadata):
- * ext/speex/gstspeexdec.c: (speex_dec_chain):
- * ext/theora/theoradec.c: (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Add codec-name metadata.
+2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
-2004-10-25 Takao Fujiwara <Takao.Fujiwara@Sun.COM>
+ * common:
+ * gst-plugins-base.spec.in:
+ update spec file for RTP changes
+ Original commit message from CVS:
+ update spec file for RTP changes
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- * ext/alsa/gstalsamixertrack.h:
- * po/POTFILES.in:
- ALSA mixer track label internationalization (#154054).
+ gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (add_sink),
+ (gst_play_bin_change_state):
+ Activate sync in playbin, we are ready to handle it for live streams.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/theoradec.c: (theora_dec_chain):
- Export bitrate as metadata.
+ tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
+ Original commit message from CVS:
+ * tests/check/elements/playbin.c:
+ (test_sink_usage_video_only_stream), (playbin_suite):
+ Add small test for stream-info-value-array code paths.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- * ext/alsa/gstalsamixertrack.h:
- Fix names, fix loop.
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_skew_slaving):
+ Don't try to create invalid calibration parameters by making the
+ internal time go backwards, instead make external time go forward.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * ext/speex/gstspeexdec.c: (gst_speex_dec_init),
- (speex_dec_convert):
- sinkconvert function so oggdemux can get the file length (totem).
+ gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst/playback/gstplaybasebin.c: (add_stream):
+ Fix leak in add_stream(), when g_value_set_object() increases the
+ refcount of streaminfo object. Fixes #426250.
-2004-10-25 James Morrison <ja2morri@csclub.uwaterloo.ca>
+2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ Add a test pattern called "circular", which has concentric
+ rings with varying radial frequency. The main purpose of this
+ pattern is to test fidelity loss in a filter or scaler element.
+ Notably, this pattern is scale invariant, and is optimally viewed
+ with a width (and height) of 400.
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
- Don't push incomplete packets.
- * gst/typefind/gsttypefindfunctions.c: (m4a_type_find):
- Fix MPEG-4 audio typefinding.
+2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
+ (deactivate_free_recursive):
+ Decodebin2 doesn't unref pads it obtains in some occasions:
+ - multiqueue src pads, when either connecting further or exposing
+ - sink pads of new autoplugged elements
+ - peer pads when recursively freeing elements
+ Fixes #425455.
- * sys/v4l/Makefile.am:
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_type),
- (gst_v4lelement_init), (gst_v4lelement_dispose),
- (gst_v4lelement_change_state):
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lxoverlay.c: (gst_v4l_xoverlay_open),
- (gst_v4l_xoverlay_close), (idle_refresh),
- (gst_v4l_xoverlay_set_xwindow_id):
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/v4l-overlay_calls.c:
- * sys/v4l/v4l_calls.h:
- * sys/v4l2/Makefile.am:
- * sys/v4l2/gstv4l2.c: (plugin_init):
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_get_type),
- (gst_v4l2element_init), (gst_v4l2element_dispose),
- (gst_v4l2element_change_state):
- * sys/v4l2/gstv4l2element.h:
- * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open),
- (gst_v4l2_xoverlay_close), (idle_refresh),
- (gst_v4l2_xoverlay_set_xwindow_id):
- * sys/v4l2/gstv4l2xoverlay.h:
- * sys/v4l2/v4l2-overlay_calls.c:
- * sys/v4l2/v4l2_calls.h:
- Remove client-side overlay handling, use the X-server v4l plugin
- for that. Nicer overlay, less code. Also make the plugin
- compileable without X (but then without overlay, obviously).
- Makes xwindowlistener obsolete, should we remove that?
+2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Add audio/x-raw-float support, now that audioconvert support
+ non-native endianness floats.
- * sys/oss/gstosssrc.c: (gst_osssrc_get_time), (gst_osssrc_get),
- (gst_osssrc_src_query):
- * sys/oss/gstosssrc.h:
- OK, so people want offset in DEFAULT. This time, actually fix all
- cases.
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Add FPS properly.
+2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ gstreamer-plugins-base.pc doesn't exist, it's
+ gstreamer-plugins-base-0.10.pc.
- * gst/asfdemux/gstasfmux.c:
- * gst/avi/gstavimux.c:
- Framerate.
+2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
+
+ with some minor changes
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler dot de>
+ with some minor changes
+ * gst-libs/gst/floatcast/floatcast.h:
+ Use more efficient float endianness conversion functions that don't
+ involve 2 function calls per value.
+ * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
+ (check_default), (audio_convert_prepare_context):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_parse_caps), (make_lossless_changes):
+ Support non-native endianness floats as input and output.
+ Fixes #339838.
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST):
+ Add unit tests for the non-native endianness float conversions.
+
+2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_base_init),
+ (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state),
+ (gst_base_rtp_depayload_set_property),
+ (gst_base_rtp_depayload_get_property):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Add Private structure.
+ Bring element code to 2007.
+ Parse clock-base caps param and use it when generating the
+ newsegment.
+ Reset variables before going to PAUSED.
+ Fix some docs.
+
+2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add RTCP docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_get_adapter):
+ Add RTCP docs.
+ Fix some more docs.
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
+ (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
+ (gst_rtcp_buffer_get_packet_count), (read_packet_header),
+ (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
+ (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
+ (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
+ (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
+ (gst_rtcp_packet_sr_get_sender_info),
+ (gst_rtcp_packet_sr_set_sender_info),
+ (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
+ (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
+ (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
+ (gst_rtcp_packet_sdes_get_chunk_count),
+ (gst_rtcp_packet_sdes_first_chunk),
+ (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
+ (gst_rtcp_packet_sdes_first_item),
+ (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
+ (gst_rtcp_packet_bye_get_ssrc_count),
+ (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
+ (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
+ (gst_rtcp_packet_bye_get_reason_len),
+ (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Add new helper object for parsing and creating RTCP messages.
+
+2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ PCM samples with width=8 must be always unsigned, no matter what
+ depth they have.
+
+2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
+ Original commit message from CVS:
+ 2007-03-29 Andy Wingo <wingo@pobox.com>
+ * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
+ perfect offsets also, not just timestamps.
+ * tests/check/elements/videorate.c (test_more): Test that given
+ any incoming offsets, that videorate produces perfect offsets.
+
+2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add some more RIFF formats.
+
+2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_default_clock_rate):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Fix fixed payload names and docs.
+ Added method to get the default clock rates of fixed payload types.
+ API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
+
+2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
+
+ tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
+ Original commit message from CVS:
+ * tests/check/pipelines/.cvsignore:
+ Add new vorbisdec test to cvsignore.
+
+2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
+ (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
+ (gst_base_audio_sink_set_property),
+ (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
+ (clock_convert_external), (gst_base_audio_sink_resample_slaving),
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_async_play):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Store private stuff in GstBaseAudioSinkPrivate.
+ Add configurable clock slaving modes property.
+ API:: GstBaseAudioSink::slave-method property
+ Some more latency reporting tweaks.
+ Added skew based clock slaving correction and make it the default until
+ the resampling method is more robust.
+
+2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c:
+ Add docs to the integer pack functions and implement proper
+ rounding. Before we had rounding towards negative infinity, i.e.
+ always the smaller number was taken. Now we use natural rounding,
+ i.e. rounding to the nearest integer and to the one with the largest
+ absolute value for X.5. The old rounding introduced some minor
+ distortions. Fixes #420079
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Fix one unit test that assumed the old rounding and added unit tests
+ for checking signed/unsigned int16 <-> signed/unsigned int16 with
+ depth 8, one for signed int16 <-> unsigned int16 and one for the new
+ rounding from signed int32 to signed/unsigned int16.
+
+2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
+ (gst_audio_convert_transform_caps):
+ Fix typo in debug line introduced recently, as pointed out on irc.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_set_property):
- Fix properties (channel, norm, frequency).
+ Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Make sure we parse floating-point numbers in vorbis comments
+ correctly with either '.' or ',' as separator, no matter what
+ the current locale is. Add unit test for this too.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_get_property):
- Flag typo.
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_set_defaults):
- No warnings.
+ * tests/check/pipelines/vorbisdec.c:
+ commit new file
+ Original commit message from CVS:
+ commit new file
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_clear_format_list):
- Fix hang.
+ gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler de>
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
+ When writing out floating-point numbers to vorbis comment tags, always
+ use the same character as separator no matter what the current locale is
+ (fixes #423051).
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Add unit tests for replaygain tags in vorbis comments (closes #423055).
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/v4l2/gstv4l2element.h:
- Yet Another Hack (tm) for kernel header borkedness.
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
- (gst_v4l2src_v4l2fourcc_to_caps), (gst_v4l2_fourcc_from_structure),
- (gst_v4l2src_link), (gst_v4l2src_getcaps),
- (gst_v4l2src_change_state):
- * sys/v4l2/gstv4l2src.h:
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_init),
- (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop):
- Fix caps, keep track of state, work.
+ ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
+ vorbis_handle_data_packet):
+ Correctly set DURATION to generate a timestamp-continuous stream.
+ One bug left at the end; see
+ ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
+ Add a test to check this. Without the above patch this test fails.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Quiet.
+ gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/Makefile.am:
+ The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Don't mix bytes and samples.
+ * gst-plugins-base.spec.in:
+ update spec file
+ Original commit message from CVS:
+ update spec file
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
- * ext/ogg/gstoggmux.c:
- Basic pad template which accepts OGM tracks, speex, flac, vorbis
- and theora. Any is incorrect.
- * gst/asfdemux/gstasfmux.c: (gst_asfmux_vidsink_link):
- Fix caps.
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_base_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_base_init),
- (gst_v4lmjpegsrc_init), (gst_v4lmjpegsrc_srcconnect),
- (gst_v4lmjpegsrc_getcaps), (gst_v4lmjpegsrc_change_state):
- * sys/v4l/gstv4lmjpegsrc.h:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_src_link), (gst_v4lsrc_getcaps),
- (gst_v4lsrc_change_state):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_capture_start),
- (gst_v4lmjpegsrc_capture_stop):
- Fix caps. Keep track of internal state. Work.
-
-2004-10-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
+ (gst_video_rate_reset), (gst_video_rate_chain):
+ If videorate changes caps, we can no longer use the old buffer
+ (which may have a different size, incompatible with our caps).
+ So don't do that; just duplicate the new frame more times.
- * ext/Makefile.am:
- Fix the build fixes.
+2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
+ Remove playbin's override of the set_clock vmethod. It's irrelevant
+ after Wim's commit on the 19th.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (_find_chain_seek),
- (gst_ogg_pad_push):
- Check for pad availability before using it.
- * ext/ogg/gstoggdemux.c: (_find_chain_process):
- Fix parsing of chained ogg. Needs more work on the decoder side.
+2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
+ Original commit message from CVS:
+ * gst-libs/gst/app/Makefile.am:
+ Use GST_ALL_LDFLAGS, which actually exists, but maybe David
+ can confirm that was what he wanted.
- * gst/spectrum/Makefile.am:
- * gst/spectrum/demo-osssrc.c: (spectrum_chain), (main),
- (idle_func):
- Fix demo and reenable it. Yes, I'm currently playing with audio
- analysis tools
+2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
+ (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ Don't cache file sizes. Fixes #341078.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- We love it if files that start at zero work too...
+2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (add_sink):
+ Use GST_PTR_FORMAT to log caps.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate):
- Handle files with missing EOS headers.
+2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-10-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/subparse/samiparse.c: (handle_start_font):
+ Special-case some more colour names that pango doesn't handle by
+ default. Fixes #420578.
- * gst/tcp/gsttcpserversink.c:
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_init_send):
- Zero some variables first (need for accept not to return EINVAL)
+2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
+ If we get a zero-sized input buffer, don't pass it to libvorbis, as
+ that marks EOS internally. After that, libvorbis will buffer all
+ input data, and encode none of it, eventually leading to memory
+ exhaustion.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_push):
- * ext/theora/theoradec.c: (theora_dec_sink_convert),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (gst_vorbis_dec_init), (vorbis_dec_convert), (vorbis_dec_chain):
- Seeking and querying finetune.
+2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (remove_fakesink):
+ Don't post STATE_DIRTY anymore.
+ * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
+ (gst_play_bin_change_state):
+ Remove stream_time reset in seek handling, core does that now.
+ Disable clocking for live pipelines by forcing a NULL clock to the
+ complete pipeline, core is too smart now for our previous hack.
+ We can always autoplug in PAUSED now.
- * configure.ac:
- * ext/Makefile.am:
- * ext/raw1394/Makefile.am:
- fix the build
+2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
+ Original commit message from CVS:
+ * REQUIREMENTS: Update this file, change the formatting to make
+ it more consistent, plus more machine readable.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate):
- Wrong return.
- * gst/playback/Makefile.am:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
- Fix marshallers.
+2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
+ (strip_width_64), (append_with_other_format):
+ Previous fix was too simplistic, and broke the tests. Use a better
+ approach; only strip 64 from widths for integer audio.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event):
- Silence.
+2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
+ (gst_audio_convert_transform_caps):
+ We don't support 64 bit integer audio, so don't try to claim we can.
+ Stops us producing caps don't match our template caps.
+ Update comments.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_populate),
- (gst_ogg_pad_push):
- Yay for non-lineair granulepos in theora.
+2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
-2004-10-18 Wim Taymans <wim@fluendo.com>
+ gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ (audioresample_check_discont), (audioresample_transform):
+ Don't trigger discontinuities for very small imperfections; a filter
+ flush will sound bad, and many plugins have rounding errors leading
+ to these.
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_push), (gst_dvdec_loop):
- * ext/dv/gstdvdec.h:
- Make sure we renegotiate aspect ratio when the camera switches.
+2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ Add min-ptime property to RTP base audio payloader. Patch by
+ olivier.crete@collabora.co.uk.
+ Fixes #415001
+ Indentation/whitespace/documentation fixes.
+
+2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
+ Original commit message from CVS:
+ 2007-03-14 Julien MOUTTE <julien@moutte.net>
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
+ (audioresample_transform_size), (audioresample_do_output),
+ (audioresample_transform), (audioresample_pushthrough): Handle
+ discontinuous streams.
+ * gst/audioresample/gstaudioresample.h:
+ * tests/check/elements/audioresample.c:
+ (test_discont_stream_instance), (GST_START_TEST),
+ (audioresample_suite): Add a test for discontinuous streams.
+ * win32/common/config.h: Updated.
+
+2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ po/: Update translations from translation project.
+ Original commit message from CVS:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update translations from translation project.
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_push):
- Start at zero.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Skip headers. Bad idea for chained ogg, but fixes seeking.
+ * gst/gdp/gstgdpdepay.c:
+ add buffer logging
+ Original commit message from CVS:
+ add buffer logging
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- I swear, this is the last time I touch this.
+ gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
+ Original commit message from CVS:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/resample.c: (resample_init):
+ Since I really am not interested in a debug line for each sample
+ being processed, move the library's debugging to its own category,
+ libaudioresample
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_populate),
- (_read_bos_process), (gst_ogg_demux_iterate), (gst_ogg_pad_new):
- Faster seeking.
- * ext/theora/theoradec.c: (theora_dec_sink_convert):
- Time-to-default conversion.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Don't error on unknown packets, just skip. We should probably
- read them if we want to support chained ogg.
+ * gst/audioresample/gstaudioresample.c:
+ add debugging and reformat docs
+ Original commit message from CVS:
+ add debugging and reformat docs
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- Added cdaudio to wrong list.
+ ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_type_packet):
+ Since the plugin doesn't support anything other than 4:2:0 right
+ now, post an error and fail if we get something else. Won't matter
+ until libtheora supports the other pixel formats, but hopefully
+ that'll be soon...
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
- * configure.ac:
- Revive cdaudio.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_push):
- * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
- (gst_smokeenc_resync), (gst_smokeenc_chain):
- Fix mimetype on smoke encoder.
- Add aspect ratio to dvdec. Not sure if these
- values are correct though....
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init):
- Fix vorbis property descriptions and ranges.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate):
- Really do nothing when no data is available.
- Go to the playing state when the stream is not seekable
- instead of failing.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/cdaudio/gstcdaudio.c: (_do_init), (gst_cdaudio_base_init),
- (gst_cdaudio_get_event_masks), (gst_cdaudio_send_event),
- (gst_cdaudio_query), (plugin_init), (cdaudio_uri_get_type),
- (cdaudio_uri_get_protocols), (cdaudio_uri_get_uri),
- (cdaudio_uri_set_uri), (cdaudio_uri_handler_init):
- Added uri handler for cd://
- Port to new API.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (find_compatibles), (close_pad_link),
- (try_to_link_1), (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (gen_preroll_element),
- (remove_prerolls), (unknown_type), (add_element_stream),
- (new_decoded_pad), (setup_source), (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_link_stream):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_vis_element), (remove_sinks), (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (gst_stream_info_get_type), (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (stream_info_mute_pad),
- (gst_stream_info_set_property), (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Add sink padtemplate to decodebin.
- Added some more comments.
- Make queue size configurable in playbasebin.
- Added possibility to use elements as sinks (ex cdaudio).
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/speex/gstspeexenc.c: (gst_speexenc_class_init),
- (gst_speexenc_chain):
- Fix speex timestamps so that it gets muxed properly.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_get_type),
- (gst_dv1394src_base_init), (gst_dv1394src_class_init),
- (gst_dv1394src_init), (gst_dv1394src_dispose),
- (gst_dv1394src_iso_receive), (gst_dv1394src_discover_avc_node),
- (gst_dv1394src_change_state), (gst_dv1394src_get_event_mask),
- (gst_dv1394src_event), (gst_dv1394src_get_formats),
- (gst_dv1394src_convert), (gst_dv1394src_get_query_types),
- (gst_dv1394src_query), (gst_dv1394src_uri_get_type),
- (gst_dv1394src_uri_get_protocols), (gst_dv1394src_uri_get_uri),
- (gst_dv1394src_uri_set_uri), (gst_dv1394src_uri_handler_init):
- * ext/raw1394/gstdv1394src.h:
- Added conversion/query functions.
- Update buffer timestamps,
- Added signals.
- Added uri dv:// so it might play from the firewire in playbin.
- Fix a possible leak.
- Added debugging.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
- (gst_dv1394src_init), (gst_dv1394src_set_property),
- (gst_dv1394src_get_property), (gst_dv1394src_iso_receive),
- (gst_dv1394src_discover_avc_node), (gst_dv1394src_change_state):
- * ext/raw1394/gstdv1394src.h:
- Added AV/C VTR control support needed for some cameras.
- Added automatic port detection.
- Added properties for selecting the channel.
- The configure.ac script is not yet updated to reflect the
- new libavc1394 and librom1394 dependencies.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse), (gst_qtdemux_handle_esds):
- An esds box is not a container.
- Fix parsing of mp4v boxes.
- Do not try to renegotiate fps for each frame. Need to
- find a better method. This should fix mp4 playback.
-
-2004-10-14 David Schleef <ds@schleef.org>
-
- * configure.ac: update for swfdec-0.3 and liboil-0.2
- * ext/swfdec/gstswfdec.c: update for swfdec-0.3
- * ext/swfdec/gstswfdec.h: same
- * gst/videofilter/gstvideobalance.c: update for liboil-0.2
- * gst/videotestsrc/videotestsrc.c: same
-
-2004-10-14 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
- (is_sync_frame), (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
- Turn warnings into info.
- Don't allow a state change in the streaming thread.
-
-2004-10-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/oggvorbisenc.c:
- * ext/vorbis/vorbisdec.c:
- fix template sample rate
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ Mention Patch by: Alex Lancaster in a recent commit.
+
+2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
+ Original commit message from CVS:
+ * examples/app/.cvsignore:
+ The buildbot demands .cvsignore files, and I comply.
+
+2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
+
+ Add appsrc/appsink example.
+ Original commit message from CVS:
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/app/Makefile.am:
+ * examples/app/appsrc_ex.c:
+ Add appsrc/appsink example.
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstapp.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * gst/app/gstapp.c:
+ Add appsink.
+
+2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
+ Use gst_guint64_to_gdouble for conversion.
+ * win32/MANIFEST:
+ Add new files to the win32 MANIFEST.
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstpbutils.def:
+ Add new exported functions.
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstdecodebin.dsp:
+ * win32/vs6/libgstplaybin.dsp:
+ Change the link to libgstpbutils.lib.
+ * win32/vs6/libgstdecodebin2.dsp:
+ Add a new project for decodebin2.
+ * win32/vs6/libgstpbutils.dsp:
+ Add a new project for pbutils.
+
+2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ Also accept partial dates with only year and month,
+ like 1999-12-00 (fixes #410396 even more).
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Add unit test for the above.
-2004-10-13 Wim Taymans <wim@fluendo.com>
+2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
- Decoding the header first fixes some problems in resyncing
- in more mp3s.
+ tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
+ Original commit message from CVS:
+ * tests/check/elements/subparse.c: (GST_START_TEST),
+ (subparse_suite):
+ Add unit test for MPL2 subtitle format (#413799).
-2004-10-12 Wim Taymans <wim@fluendo.com>
+2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
+
+ gst/subparse/: Add support for MPL2 subtitle format (#413799).
+ Original commit message from CVS:
+ Patch by: Kamil Pawlowski <kamilpe gmail com>
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect),
+ (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
+ (gst_subparse_type_find):
+ * gst/subparse/gstsubparse.h:
+ * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
+ * gst/subparse/mpl2parse.h:
+ Add support for MPL2 subtitle format (#413799).
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_vis_element), (remove_sinks), (setup_sinks):
- Added vis plugin support, need to configure the vis
- element to activate it.
+2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: We require core CVS for the new buffer metadata copy functions.
+ Original commit message from CVS:
+ * configure.ac:
+ We require core CVS for the new buffer metadata copy functions.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get),
- (gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
- Some debug.
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
- (gst_avi_demux_handle_src_event), (gst_avi_demux_read_superindex),
- (gst_avi_demux_read_subindexes), (gst_avi_demux_add_stream),
- (gst_avi_demux_stream_index), (gst_avi_demux_skip),
- (gst_avi_demux_sync), (gst_avi_demux_stream_scan),
- (gst_avi_demux_massage_index), (gst_avi_demux_stream_header):
- * gst/avi/gstavidemux.h:
- Support for openDML-2.0 indx/ix## chunks. Support for broken index
- recovery (where, if part of the index is broken, we will still read
- the rest of the index and recover the broken part by stream
- scanning). More broken media support. EOS workarounds. General AVI
- braindamage headache recovery. Aspirin included.
+2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
+ Fixes #414496.
+
+2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Improve adapter usage and comments.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
+ (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
+ Improve adapter usage and comments.
+
+2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Use new metadata copy function.
+ Original commit message from CVS:
+ * ext/pango/gsttextrender.c: (gst_text_render_chain):
+ * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
+ * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
+ Use new metadata copy function.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_transform):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
+ Basetransform copied the metadata for us.
+
+2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
+ (gst_text_overlay_video_event):
+ Some more logging. Only accept newsegment events in TIME format and
+ send a WARNING message if they are not in TIME format.
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
+ (gst_sub_parse_chain), (gst_sub_parse_sink_event):
+ * gst/subparse/gstsubparse.h:
+ No need to allocate GstSegment structure dynamically, just put it
+ into the instance structure; ignore newsegment events in BYTE
+ format and in particular don't let it overwrite our saved TIME
+ segment from the last seek.
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_open),
- (cdparanoia_event), (cdparanoia_query):
- Get rid of hideous lead-in.
+ gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
+ Replace AC3 typefinder with one that isn't terrible, and actually
+ works usefully.
-2004-10-11 Wim Taymans <wim@fluendo.com>
+2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Wrong var used to get g_list_next.
+ gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_transform):
+ fix error category and translatable string
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_get), (cdparanoia_open):
- Report discid as metadata, add duration.
+ pkgconfig/: Fix up utils => pbutils here too.
+ Original commit message from CVS:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ Fix up utils => pbutils here too.
-2004-10-11 Wim Taymans <wim@fluendo.com>
+2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Cleanup the previous pipeline a little earlier for the
- case that a source element provides raw data.
+ gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (handle_buffer):
+ Break out of loop in chain function as soon as possible if we get
+ a non-OK flow return.
-2004-10-11 Benjamin Otte <otte@gnome.org>
+2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
+ Original commit message from CVS:
+ * tests/check/elements/alsa.c: (GST_START_TEST):
+ Unref the mixer if the state change fails too (if the
+ alsa devices are inaccessible, for example)
+
+2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Don't test libvisual elements in the states check, because libvisual
+ seems to leak internally.
+ Re-enable the alsa and states tests now that there's new suppressions
+ in gst.supp.
+ * tests/check/elements/alsa.c: (GST_START_TEST):
+ Don't leak the alsamixer we instantiated.
+
+2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
+ (gst_ximagesink_change_state), (gst_ximagesink_reset),
+ (gst_ximagesink_finalize):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
+ (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
+ Move some cleanup stuff from the state change handler into a _reset()
+ function that can be called from _finalize(). This ensures that things
+ get freed even if (for some reason) the NULL->READY state transition
+ fails in the parent class.
+ Even if a parent state change fails, process our downward state change
+ logic instead of bailing out early.
+ Free the correct xcontext pointer in ximagesink's xcontext_clear.
+
+2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/alsa/gstalsasink.c: Extra log line.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open):
+ Extra log line.
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
+ * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
+ Use pango_font_description_set_family_static instead of
+ pango_font_description_set_family to save a string copy (it was
+ leaking due to the strdup anyway)
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
+ * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
+ Chain up in finalize.
+
+2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ (gst_mixer_track_class_init), (gst_mixer_track_get_property),
+ (gst_mixer_track_set_property):
+ API: add "untranslated-label" property which should be set by
+ implementations at construct time (#414645).
+ * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
+ Set "untranslated-label" when constructing mixer track objects.
+ * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
+ Unit test to check the above.
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- reset v1 tag offset when there is no v1 tag. Fixes id3demux always
- consuming the last 128 bytes, even though it was valid mp3 data.
+2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ ext/ogg/gstoggdemux.c: Fix confusing debug message.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
+ Fix confusing debug message.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_get):
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- Change g_warnings to GST_WARNING_OBJECT and fix colourspace issue
+2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst-plugins-base.doap: update doap file with new version
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ update doap file with new version
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_src_link), (gst_v4lsrc_getcaps):
- Fix for webcams that support only specific width or height
+2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+ * gst/tcp/gstmultifdsink.c:
+ update docs
+ Original commit message from CVS:
+ update docs
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
- Fix wrong discont event setup (fixes #154967).
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
-2004-10-09 Sebastien Cote <sc5@hermes.usherb.ca>
+=== release 0.10.12 ===
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Error out on invalid data (fixes #154807).
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ Release 0.10.12
+ Original commit message from CVS:
+ Release 0.10.12
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * common:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
- * ext/dvdread/dvdreadsrc.c: (_read):
- Make titles > 0 work again (fixes #154834).
+2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Bump version to 0.10.11.4 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version to 0.10.11.4 pre-release
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_template_caps):
- WMV3 missing in template caps.
+2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_async_play):
+ Fix regression that made GStreamer skip the first samples of audio.
+ Fixes #414684.
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- OK, so the original code was too strict. It makes random AVI files
- hang for seconds upon opening, which is unacceptable and is far
- beyond the original goal of getting multiple chunks for one-chunk
- sounc stream files. So now do just that.
+2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Bump version to 0.10.11.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version to 0.10.11.3 pre-release
- * gst/playback/gstplaybasebin.c: (setup_source),
- (gst_play_base_bin_change_state):
- Actually clean up streaminfo if output fails. This would trigger
- if, for example, there was no CD in the drive. No preroll, so
- a streaminfo structure is created, but the subsequent state change
- of the thread fails.
- * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
- Don't change state if parent failed.
+2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-10-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Update paths for the rename from utils to pbutils to fix the build.
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_init), (gst_play_bin_get_property), (handoff),
- (gen_video_element), (remove_sinks):
- Add small bits of code for screenshot handling.
+2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/Makefile.am:
+ Change directory to install headers in from gst/utils to gst/pbutils
+ as well.
- * gst/playback/gstplaybin.c: (gst_play_bin_set_property),
- (gen_video_element), (gen_audio_element), (setup_sinks):
- Don't assume the user provided sinks are named "sink"...
+2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ * tests/check/libs/.gitignore:
+ moap ignore
+ Original commit message from CVS:
+ moap ignore
- * gst/playback/gstplaybasebin.c: (gen_preroll_element),
- (unknown_type), (setup_source), (gst_play_base_bin_remove_element),
- (gst_play_base_bin_link_stream):
- Do not try to autoplug sources that generate raw streams like
- cdparanoia.
- disconnect the preroll overrun signal when we don't need it anymore.
+2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-08 Milosz Derezynski <internalerror.rez@fhtw-berlin.de>
+ * win32/common/config.h:
+ * win32/common/libgstutils.def:
+ update defs
+ Original commit message from CVS:
+ update defs
- * ext/cdparanoia/gstcdparanoia.c: (_do_init),
- Added reworked patch from #154903 from milosz derezynski (deadchip).
+2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ rename utils to pbutils
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/pbutils/descriptions.c:
+ (gst_pb_utils_get_source_description),
+ (gst_pb_utils_get_sink_description),
+ (gst_pb_utils_get_decoder_description),
+ (gst_pb_utils_get_encoder_description),
+ (gst_pb_utils_get_element_description),
+ (gst_pb_utils_add_codec_description_to_tag_list),
+ (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
+ * gst-libs/gst/pbutils/descriptions.h:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/pbutils/install-plugins.h:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ (gst_missing_uri_source_message_new),
+ (gst_missing_uri_sink_message_new),
+ (gst_missing_element_message_new),
+ (gst_missing_decoder_message_new),
+ (gst_missing_encoder_message_new),
+ (gst_missing_plugin_message_get_description):
+ * gst-libs/gst/pbutils/missing-plugins.h:
+ * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
+ * gst-libs/gst/pbutils/pbutils.h:
+ * gst-libs/gst/utils/Makefile.am:
+ * gst-libs/gst/utils/base-utils.c:
+ * gst-libs/gst/utils/base-utils.h:
+ * gst-libs/gst/utils/descriptions.c:
+ * gst-libs/gst/utils/descriptions.h:
+ * gst-libs/gst/utils/install-plugins.c:
+ * gst-libs/gst/utils/install-plugins.h:
+ * gst-libs/gst/utils/missing-plugins.c:
+ * gst-libs/gst/utils/missing-plugins.h:
+ * gst-plugins-base.spec.in:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybasebin.c: (setup_subtitle),
+ (gen_source_element):
+ * gst/playback/gstplaybin.c: (plugin_init):
+ * tests/check/Makefile.am:
+ * tests/check/libs/pbutils.c: (GST_START_TEST),
+ (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
+ * tests/check/libs/utils.c:
+ rename utils to pbutils
- * ext/cdparanoia/gstcdparanoia.c: (_do_init),
- (cdparanoia_base_init), (cdparanoia_class_init), (cdparanoia_init),
- (cdparanoia_dispose), (cdparanoia_get), (cdparanoia_change_state),
- (cdparanoia_convert), (cdparanoia_uri_get_type),
- (cdparanoia_uri_get_protocols), (cdparanoia_uri_get_uri),
- (cdparanoia_uri_set_uri), (cdparanoia_uri_handler_init):
- * ext/cdparanoia/gstcdparanoia.h:
- This adds the cdda://<tracknum> uri.
+2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/app/Makefile.am: Install the headers.
+ Original commit message from CVS:
+ * gst-libs/gst/app/Makefile.am:
+ Install the headers.
+
+2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
+ Original commit message from CVS:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstappbuffer.c:
+ * gst-libs/gst/app/gstappbuffer.h:
+ * gst-libs/gst/app/gstappsrc.c:
+ Add GstAppBuffer that includes a callback and closure for
+ proper handling of data chunks.
+
+2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ Hacking to address issues in 413418.
+
+2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
+
+ Move the app library to gst-libs/gst/app (duh!)
+ Original commit message from CVS:
+ * Makefile.am:
+ * configure.ac:
+ * ext/Makefile.am:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstapp.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * gst/app/Makefile.am:
+ * gst/app/gstapp.c:
+ * gst/app/gstappsrc.c:
+ * gst/app/gstappsrc.h:
+ Move the app library to gst-libs/gst/app (duh!)
+
+2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Add documentation for decodebin2 that indicates that the API is still unstable.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
+ Add documentation for decodebin2 that indicates that the API
+ is still unstable.
+
+2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
+ Original commit message from CVS:
+ * configure.ac:
+ Update to 0.10.11.2 (0.10.12 pre-release)
+
+2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_async_play):
+ base time is irrelevant here.
+
+2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/: Improve debugging.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
+ * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
+ Improve debugging.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_query), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
+ Improve latency and clock slaving calculations.
+ Improve slave clock calibration.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full):
+ When we are asked to render N sample to 0 bytes, return N.
+
+2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.*: Remove unused dispose function.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
+ (gst_alsasink_write), (gst_alsasink_reset):
+ * ext/alsa/gstalsasink.h:
+ Remove unused dispose function.
+ Rename lock to not interfere with alsasrc lock.
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
+ (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
+ (gst_alsasrc_read), (gst_alsasrc_reset):
+ * ext/alsa/gstalsasrc.h:
+ Implement finalize function.
+ Use lock to protect alsa access.
+ Implement _reset.
+ Fine tune sw params.
+
+2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * common:
+ * configure.ac:
+ typo
+ Original commit message from CVS:
+ typo
+
+2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ configure.ac: Convert to new AG_GST style.
+ Original commit message from CVS:
+ * configure.ac:
+ Convert to new AG_GST style.
-2004-10-08 Wim Taymans <wim@fluendo.com>
+2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
+
+ gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
+ Original commit message from CVS:
+ Patch by: Ed Catmur <ed at catmur dot co dot uk>
+ * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
+ (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
+ Fix race condition when rapidly switching visualisations in playbin.
+ Fixes #401029.
+
+2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Include local stuff before system installed things in LDFLAGS and
+ CFLAGS.
+
+2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Improve debugging.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
+ Improve debugging.
+
+2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ sys/v4l/: Fix duration and timestamping, taking latency into account.
+ Original commit message from CVS:
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
+ (gst_v4lsrc_fixate), (gst_v4lsrc_query):
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
+ Fix duration and timestamping, taking latency into account.
+ Implement latency query.
+
+2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
+ (gst_audio_clock_new):
+ Fix clock name.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_init), (gst_base_audio_sink_query):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
+ (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
+ (gst_base_audio_src_create):
+ Improve latency query code.
+ Use proper clock names.
+
+2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (find_compatibles), (close_pad_link),
- (try_to_link_1), (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (unknown_type), (gst_play_base_bin_remove_element),
- (gst_play_base_bin_link_stream):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_init),
- (gst_play_bin_set_property), (gen_video_element),
- (gen_audio_element), (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (gst_stream_info_get_type), (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (stream_info_mute_pad),
- (gst_stream_info_set_property), (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Reuse the audio and video bins.
- Some internal cleanups in the stream selection code.
+ * tests/check/generic/states.c:
+ plug test leak
+ Original commit message from CVS:
+ plug test leak
-2004-10-08 Julien MOUTTE <julien@moutte.net>
+2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h: Reverting Ronald's changes as the issue is
- not coming from those elements. Moreover these elements should not keep
- the xid they have been given when in NULL state.
+ tests/check/generic/states.c: Copy the states.c test from core again
+ Original commit message from CVS:
+ * tests/check/generic/states.c: (GST_START_TEST):
+ Copy the states.c test from core again
+ * tests/check/Makefile.am:
+ ignore cdio and cdparanoiasrc
-2004-10-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h:
- Actually only create a new toplevel window if we're not gonna
- embed it right after.
+ gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
+ (double_hq), (audio_convert_get_func_index), (check_default),
+ (audio_convert_prepare_context), (audio_convert_convert):
+ Also make valgrind happy and avoid copying data in some cases.
-2004-10-07 Wim Taymans <wim@fluendo.com>
+2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybasebin.c: (play_base_bin_mute_pad),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream):
- * gst/playback/gstplaybin.c: (setup_sinks):
- Implement muting/unmuting of streams, mute streams that are not
- used.
+ * tests/check/generic/states.c:
+ use a macro
+ Original commit message from CVS:
+ use a macro
+
+2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
+ (double_hq), (audio_convert_get_func_index),
+ (audio_convert_prepare_context), (audio_convert_convert):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
+ (gst_audio_convert_transform_caps):
+ * tests/check/elements/audioconvert.c: (GST_START_TEST),
+ (audioconvert_suite):
+ Don't run inplace if that overwrites source data as we go. Add more
+ tests. Fixes #339837 even more.
-2004-10-07 Wim Taymans <wim@fluendo.com>
+2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
+ Original commit message from CVS:
+ 2007-02-27 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
+ (msg_segment_done): Fix various seeking bugs (Slider was not
+ updating when doing a non flushing seek, Reverse playback
+ on segment seek was wrong).
+
+2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
+
+ Add a new plugin/library to make it easy for apps to shove data into a pipeline.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/app/Makefile.am:
+ * gst/app/gstapp.c:
+ * gst/app/gstappsrc.c:
+ * gst/app/gstappsrc.h:
+ Add a new plugin/library to make it easy for apps to shove
+ data into a pipeline.
+
+2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (stop_seek):
+ When we stop scrubbing, don't leave the pipeline PLAYING when we
+ requested a PAUSED state.
+
+2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
+
+ gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler de>
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ Parse date strings in vorbis comments that have an invalid (zero)
+ month or day (#410396).
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Test case for the above.
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find),
- (plugin_init):
- Added lame audio/x-ac3 typefind function.
+2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
+ Original commit message from CVS:
+ Patch by: Loïc Minier <lool+gnome at via ecp fr>
+ * configure.ac:
+ * ext/alsa/Makefile.am:
+ * gst/audiotestsrc/Makefile.am:
+ Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
- * configure.ac:
- bump nano to cvs
+2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.8.5 ===
+ gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ Improve docs: point out that the application needs to assist playbin
+ with buffering.
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.5, "Take You On"
+ Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
+ Original commit message from CVS:
+ * gst-libs/gst/utils/install-plugins.c:
+ * gst-libs/gst/utils/missing-plugins.c:
+ * tests/check/libs/utils.c: (missing_msg_check_getters):
+ Change GStreamer marker prefix in detail string from 'gstreamer.net'
+ to just 'gstreamer'. Document the caps string component of the
+ decoder/encoder detail a bit better, since not everyone will be
+ familiar with the GStreamer media type/caps system (but they better
+ enjoy nested itemized lists).
-2004-10-06 Wim Taymans <wim@fluendo.com>
+2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- * gst/playback/gstplaybin.c: (gen_video_element):
- Do not signal the no_more_pads after the first pad when
- we are plugging a non dynamic element with multiple
- output pads (like swfdec, dvdec, ...).
+ gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
+ Original commit message from CVS:
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
+ Fix copying of GstNetBuffer (would crash before, or at least lead to
+ invalid memory access, #410772), for now by copying the GstBuffer copy
+ code from the core over here so we can copy the GstBuffer fields on a
+ provided buffer instance (of type GstNetBuffer in this case). Would be
+ better to fix this with some support by the core though (and in the long
+ run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
+ * tests/check/Makefile.am:
+ Enable unit test for GstNetBuffer.
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
- * configure.ac:
- bump for prerelease
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ 2007-02-22 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_init): Disable pull-mode activation until we
+ figure out how to make audio sinks go to PLAYING.
+
+2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
+ (double_hq), (audio_convert_get_func_index),
+ (audio_convert_prepare_context), (audio_convert_convert):
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
+ (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
+ * gst/audioconvert/gstchannelmix.h:
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Add float as an intermediate format, as well as float mixing. Enable
+ test that was failing before. Fixes #339837
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/wavparse/gstwavparse.c:
- add ATRAC3 to STATIC CAPS to fix a warning
+ tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (do_seek):
+ Undo the previous commit: -1 as a stop time implies that the stop
+ time is the end of file, clearing any previously configured segment.
- * gst/matroska/ebml-read.c:
- * gst-libs/gst/riff/riff-read.c:
- fix typos
+2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (do_seek):
+ Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
- * gst-libs/gst/riff/riff-media.c:
- generate caps for ATRAC3 audio streams
+2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/realmedia/rmdemux.c:
- generate caps for ATRAC3 audio streams
+ gst/volume/gstvolume.c: Unbreak volume, value remains gint.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_process_int16),
+ (volume_process_int16_clamp), (volume_set_caps):
+ Unbreak volume, value remains gint.
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/wavparse/Makefile.am
- * gst/wavparse/riff.h
- * gst/wavparse/wavparse.vcproj
- riff.h removal (unused and duplication with riff-ids.h)
+ gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (volume_update_real_volume), (gst_volume_set_volume),
+ (gst_volume_init), (volume_process_double), (volume_process_float),
+ (volume_process_int16), (volume_process_int16_clamp),
+ (volume_set_caps), (volume_transform_ip), (volume_update_volume):
+ * gst/volume/gstvolume.h:
+ Extend float audio support (double) and some int->uint cleanups.
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
+ (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
+ (sort_end_pads), (gst_decode_group_expose),
+ (gst_decode_group_hide):
+ Don't free groups from the streaming threads. Just put them aside and
+ free them in dispose.
+
+2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (connect_element),
+ (pad_added_group_cb), (gst_decode_group_check_if_blocked),
+ (sort_end_pads), (gst_decode_group_expose):
+ Handle dynamic pads within groups.
+ Sort pads before exposing them in order to make playbin happy.
+ There still is a race with the multiqueue filling up. This should be
+ solved separately.
+ Fixes #398721
+
+2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
+ Original commit message from CVS:
+ * gst-libs/gst/utils/base-utils.c:
+ * gst-libs/gst/utils/descriptions.c:
+ * gst-libs/gst/utils/install-plugins.c:
+ * gst-libs/gst/utils/missing-plugins.c:
+ Some more docs (and descriptions for two subtitle formats).
- * gst/wavparse/gstwavparse.h
- remove duplicated defines for audio codec codes
+2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-ids.h
- * gst/wavenc/riff.h:
- add "4CC" code for ATRAC3 audio streams
- add "4CC" code for ITU_G721_ADPCM (unused for now)
+ gst-libs/gst/audio/audio.c: Fix documentation.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Fix documentation.
-2004-10-06 Wim Taymans <wim@fluendo.com>
+2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
- * gst/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_loop):
- Actually _do_ negotiation. Pass gdouble as arg instead
- of guint64 for the framerate.
+ gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
+ Original commit message from CVS:
+ Patch by: Yves Lefebvre <ivanohe abacom com>
+ * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
+ Don't leak caps. Fixes #408278.
-2004-10-06 Wim Taymans <wim@fluendo.com>
+2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (find_compatibles), (close_pad_link), (try_to_link_1),
- (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- Set state on newly added element to READY so that negotiation
- can happen ASAP.
- Addes some more debug info.
- Do not try to plug pads with multiple caps structures or ANY
- because it is too dangerous since we do not do dynamic
- replugging.
+ More docs coverage and some ChangeLog surgery (add missing names)
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ * ext/ogg/gstoggdemux.h:
+ * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
+ (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
+ (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/interfaces/videoorientation.h:
+ * gst/adder/gstadder.h:
+ More docs coverage and some ChangeLog surgery (add missing names)
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
- written by: Gora Mohanty <gora_mohanty@yahoo.co.in>
+ sys/: Small constifications.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c:
+ (gst_ximagesink_calculate_pixel_aspect_ratio):
+ * sys/xvimage/xvimagesink.c:
+ (gst_xvimagesink_calculate_pixel_aspect_ratio):
+ Small constifications.
- * po/LINGUAS:
- * po/or.po:
- add Oriya translation
+2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
+ (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_change_state):
+ Answer latency query.
+ Use configured latency when syncing.
+ Fix clock slaving.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
+ (gst_base_audio_src_query), (gst_base_audio_src_change_state):
+ Fix possible memleak.
+ Implement latency query.
+ Small cleanups.
+
+2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
+ Ignore errors in reset, these are not fatal. They also grab the element
+ lock which is already taking when this function is called. Fixes
+ #405451.
+
+2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- Prevent overwrite of size member. Makes audio sound crappy.
+ * gst-plugins-base.spec.in:
+ add header file for easy codec install
+ Original commit message from CVS:
+ add header file for easy codec install
-2004-10-05 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add rmvb to the list of known RealMedia extensions
+ configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
+ Original commit message from CVS:
+ * configure.ac:
+ Remove 'tests/examples/xerror/Makefile' from output files again.
-2004-10-05 Wim Taymans <wim@fluendo.com>
+2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/libmng/gstmngdec.c: (gst_mngdec_loop), (mngdec_error),
- (mngdec_openstream), (mngdec_closestream),
- (mngdec_handle_sink_event), (mngdec_readdata),
- (mngdec_gettickcount), (mngdec_settimer), (mngdec_processheader),
- (mngdec_getcanvasline), (mngdec_refresh),
- (gst_mngdec_change_state):
- Set the framerate correctly.
+ Also crossref against gst-plugins-base-libs.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ Also crossref against gst-plugins-base-libs.
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- There was something wrong with the index massaging.
+ Add crossreferences to glib/gobject/gstream docs.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Add crossreferences to glib/gobject/gstream docs.
+ * gst-libs/gst/audio/audio.h:
+ Source formatting.
+ * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
+ Add own debug category.
-2004-10-04 Wim Taymans <wim@fluendo.com>
+2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
- * ext/jpeg/gstjpeg.c: (smoke_type_find), (plugin_init):
- * ext/jpeg/gstsmokedec.c: (gst_smokedec_init),
- (gst_smokedec_chain):
- * ext/jpeg/gstsmokedec.h:
- * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
- (gst_smokeenc_init), (gst_smokeenc_resync), (gst_smokeenc_chain):
- * ext/jpeg/gstsmokeenc.h:
- * ext/jpeg/smokecodec.c: (smokecodec_encode_new),
- (smokecodec_decode_new), (smokecodec_info_free),
- (smokecodec_set_quality), (smokecodec_get_quality),
- (smokecodec_set_threshold), (smokecodec_get_threshold),
- (smokecodec_set_bitrate), (smokecodec_get_bitrate),
- (find_best_size), (abs_diff), (put), (smokecodec_encode_id),
- (smokecodec_encode), (smokecodec_parse_id),
- (smokecodec_parse_header), (smokecodec_decode):
- * ext/jpeg/smokecodec.h:
- * ext/jpeg/smokeformat.h:
- Updated smoke, new bitstream, allows embedding in ogg.
+ gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler de>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
+ (#403597).
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
- Fix seeking in some files. All this code is no longer needed (and
- actually breaks stuff) because we now synchronize the full index
- right when reading the header.
+ gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ When we have external subtitles and wait for the subtitle decodebin
+ to get up and running, we set up a (sync) bus handler for the
+ subtitle decodebin, so we can stop waiting when it posts an error
+ message. However, we should do that before we set the subtitle
+ decodebin's state to playing, otherwise things are racy and we might
+ miss error messages posted before we had a chance to set up the bus.
+ This should finally fix totem hanging on .txt pseudo-subtitle files.
-2004-10-04 Wim Taymans <wim@fluendo.com>
+2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
- * configure.ac:
- configure update for libmng.
-
-2004-10-04 Wim Taymans <wim@fluendo.com>
-
- * ext/libmng/Makefile.am:
- * ext/libmng/gstmng.c: (plugin_init):
- * ext/libmng/gstmng.h:
- * ext/libmng/gstmngdec.c: (gst_mngdec_get_type),
- (gst_mngdec_base_init), (gst_mngdec_class_init),
- (gst_mngdec_sinklink), (gst_mngdec_init), (gst_mngdec_src_getcaps),
- (gst_mngdec_loop), (gst_mngdec_get_property),
- (gst_mngdec_set_property), (mngdec_error), (mngdec_openstream),
- (mngdec_closestream), (mngdec_handle_sink_event),
- (mngdec_readdata), (mngdec_gettickcount), (mngdec_settimer),
- (mngdec_processheader), (mngdec_getcanvasline), (mngdec_refresh),
- (gst_mngdec_change_state):
- * ext/libmng/gstmngdec.h:
- * ext/libmng/gstmngenc.c: (gst_mngenc_get_type),
- (mng_caps_factory), (raw_caps_factory), (gst_mngenc_base_init),
- (gst_mngenc_class_init), (gst_mngenc_sinklink), (gst_mngenc_init),
- (gst_mngenc_chain), (gst_mngenc_get_property),
- (gst_mngenc_set_property):
- * ext/libmng/gstmngenc.h:
- Added basic MNG decoder. Needs more work. The encoder does
- not work yet.
-
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/realmedia/rmdemux.c: (gst_rmdemux_handle_sink_event),
- (gst_rmdemux_loop), (gst_rmdemux_add_stream),
- (gst_rmdemux_parse_mdpr), (gst_rmdemux_dump_mdpr):
- Don't hang on length=0 chunks. Some negotiation fixes. Signal
- no-more-pads.
-
-2004-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
+ Use gst_gdouble_to_guint64 for conversions.
+ * win32/common/config.h.in:
+ Add a define for GST_INSTALL_PLUGINS_HELPER
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstcdda.def:
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstrtp.def:
+ * win32/common/libgutils.def:
+ Add new exported functions.
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstdecodebin.dsp:
+ * win32/vs6/libgstnetbuffer.dsp:
+ * win32/vs6/libgstplaybin.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgstvorbis.dsp:
+ * win32/vs6/libgstcdda.dsp:
+ * win32/vs6/libgstgdp.dsp:
+ * win32/vs6/libgstutils.dsp:
+ Update and add new project files.
- * configure.ac:
- you need at least 1.0.4 of speex
+2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-04 Iain <iaingnome@gmail.com>
+ gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
+ (subrip_remove_unhandled_tags), (parse_subrip):
+ For SubRip (.srt) subtitles, ignore all markup tags we don't
+ handle (like font tags, for example).
+ * tests/check/elements/subparse.c:
+ Add test for this.
- * ext/speex/gstspeexdec.h: Revert the includes changes.
+2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/speex/gstspeexenc.[ch]: Revert the includes changes.
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
+
+2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_fakesink),
+ (gst_decode_bin_change_state):
+ * gst/playback/gstdecodebin2.c: (add_fakesink),
+ (gst_decode_bin_change_state):
+ Don't error out if there is no fakesink in the READY to NULL state
+ change, since when decodebin is re-used, we're only adding the
+ fakesink element in READY to PAUSED.
+ * tests/check/elements/decodebin.c:
+ (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
+ (decodebin_suite):
+ Minimal unit test to make sure we can use the same decodebin
+ instance twice (at least with audiotestsrc input).
-2004-09-30 Iain <iaingnome@gmail.com>
+2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/sunaudio/gstsunaudio.c (gst_sunaudiosink_open): Use the device
- found during init or set as a property instead of hardcoding /dev/audio
+ ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
+ Try to get devic-name from device string first, and from handle only
+ as fallback (seems to yield better results and is more robust
+ against buggy probing code on the application side).
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
- * gst/realmedia/rmdemux.c: (gst_rmdemux_class_init),
- (gst_rmdemux_init), (gst_rmdemux_handle_sink_event),
- (gst_rmdemux_loop), (gst_rmdemux_add_stream), (re_hexdump_bytes),
- (re_dump_pascal_string), (gst_rmdemux_dump__rmf),
- (gst_rmdemux_dump_prop), (gst_rmdemux_parse_mdpr),
- (gst_rmdemux_dump_mdpr), (gst_rmdemux_dump_indx),
- (gst_rmdemux_dump_data):
- Use debug category, fix EOS handling. filesrc ! rmdemux now
- works.
+ ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
+ Original commit message from CVS:
+ Based on patch by: Julien Puydt <julien.puydt at laposte net>
+ * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
+ (gst_alsa_find_device_name):
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
+ Improve device-name detection a bit, especially in the case where
+ the device is not actually open (#405020, #405024). Move common code
+ into gstalsa.c instead of duplicating it.
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan), (sort), (gst_avi_demux_massage_index),
- (gst_avi_demux_stream_header), (gst_avi_demux_stream_data):
- Improve allocation, cutting and sorting of the index. How takes a
- few seconds instead of minutes.
+ gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
-2004-10-03 Christophe Fergeau <teuf@gnome.org>
+2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
+ Original commit message from CVS:
+ 2007-02-06 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
+ (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_xcontext_clear),
+ (gst_xvimagesink_interface_supported),
+ (gst_xvimagesink_probe_get_properties),
+ (gst_xvimagesink_probe_probe_property),
+ (gst_xvimagesink_probe_needs_probe),
+ (gst_xvimagesink_probe_get_values),
+ (gst_xvimagesink_property_probe_interface_init),
+ (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
+ (gst_xvimagesink_init), (gst_xvimagesink_class_init),
+ (gst_xvimagesink_get_type):
+ * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
+ for XVAdaptors so that one can choose the adaptor to use with
+ gstreamer-properties.
+
+2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ Also mention that a conversion from double to float is suboptimal still.
- * gst/realmedia/rmdemux.c: (gst_rmdemux_parse_mdpr):
- fixed compilation
+2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_class_init), (gst_audio_filter_change_state):
+ Clear our formats structure and free the caps contained in it when
+ shutting down.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add wing commander format mimetype/fourccs.
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- Don't crash if some value is 0.
+2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ 2007-02-05 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_callback): Update basesink->offset so that we
+ pull monotonically increasing offsets instead of, um, seeking back
+ to 0 each time. Fixes alsasrc ! alsasink!
+
+2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c:
+ A width and height of 1 makes us crash, so increase minimum size to
+ 2x2 pixels until someone feels like fixing this (#404512).
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add DIB fourcc (raw, palettized 8-bit RGB).
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data):
- Oops, fix strf_data reading bug.
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- Use a non-NULL tag.
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
- Time for hacks. Sorry Dave. At least one quicktime movie (a
- trailer) that I've encountered contains multiple video tracks.
- One of those is the actual video track, the other are one-frame
- tracks (images). Unfortunately, the number of frames according
- to the trak header is 1 for each, so that doesn't help. So
- instead, I look at the duration and discard tracks with a
- duration shorter than 20% of the length of the stream. Better
- than nothing.
+2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Christian Schaller <christian@fluendo.com>
+ tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
+ Add small test to make sure request pads are cleaned up properly
+ even if oggmux never changes state out of NULL.
- * ext/ivorbis/vorbis.c:
- Patch from Phil Blundell (Bug 152341)
+2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Wim Taymans <wim@fluendo.com>
+ tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
+ Original commit message from CVS:
+ * tests/check/libs/utils.c: (GST_START_TEST):
+ Fix unit test. Turns out things work much better when you
+ NULL-terminate string arrays. Should make p5 build bot happy again.
- * ext/speex/gstspeexdec.c: (gst_speex_dec_class_init),
- (speex_dec_get_formats), (speex_dec_convert),
- (speex_dec_src_query), (speex_dec_src_event), (speex_dec_event),
- (speex_dec_chain), (gst_speexdec_get_property),
- (gst_speexdec_set_property):
- Small cleanups.
+2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_base_init),
+ (gst_audio_filter_template_class_init),
+ (gst_audio_filter_template_init),
+ (gst_audio_filter_template_set_property),
+ (gst_audio_filter_template_get_property),
+ (gst_audio_filter_template_setup),
+ (gst_audio_filter_template_filter),
+ (gst_audio_filter_template_filter_inplace), (plugin_init):
+ Oops, forgot to commit fixed-up example.
-2004-10-01 Wim Taymans <wim@fluendo.com>
+2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
- (gst_wavparse_stream_init), (gst_wavparse_fmt),
- (gst_wavparse_other), (gst_wavparse_loop),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
- (gst_wavparse_srcpad_event):
- * gst/wavparse/gstwavparse.h:
- Added some more debugging info.
- Fix the case where the length of the file is 0.
- Make sure we seek to sample borders.
+ Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
+ (gst_audio_filter_class_init), (gst_audio_filter_init),
+ (gst_audio_filter_set_caps),
+ (gst_audio_filter_class_add_pad_templates):
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ Port GstAudioFilter to 0.10. This change technically breaks
+ API and ABI (and thus also every library developer's heart),
+ but seems justifiable on the grounds that the base class was
+ completely unusable before (ie. would crash immediately when
+ actually used). Fixes #403963 (and eventually also #403572).
+ Also document all of this a bit.
-2004-10-01 Wim Taymans <wim@fluendo.com>
+2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
- (gst_decode_bin_init), (find_compatibles), (close_pad_link),
- (try_to_link_1), (no_more_pads), (close_link), (type_found):
- Add some debug info to decodebin, update README
+ Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
+ Original commit message from CVS:
+ * gst-libs/gst/utils/install-plugins.c:
+ (gst_install_plugins_spawn_child):
+ * tests/check/libs/utils.c:
+ (test_base_utils_install_plugins_do_callout):
+ Lowering log level to see why things fail on the p5 build bot;
+ fix some typos in unit test messages.
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dvdnav/dvdnavsrc.c: (dvdnav_handle_navigation_event):
- Don't use g_print(); use GST_DEBUG().
+ tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
+ Original commit message from CVS:
+ * tests/check/libs/utils.c:
+ (test_base_utils_install_plugins_do_callout):
+ Don't hard-code temp directory for test helper; use GLib functions
+ to write out file and do error checking etc.
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
+ Original commit message from CVS:
+ * gst-libs/gst/utils/Makefile.am:
+ * gst-libs/gst/utils/base-utils.h:
+ * gst-libs/gst/utils/install-plugins.c:
+ (gst_install_plugins_context_set_xid),
+ (gst_install_plugins_context_new),
+ (gst_install_plugins_context_free),
+ (gst_install_plugins_get_helper),
+ (gst_install_plugins_spawn_child),
+ (gst_install_plugins_return_from_status),
+ (gst_install_plugins_installer_exited),
+ (gst_install_plugins_async), (gst_install_plugins_sync),
+ (gst_install_plugins_return_get_name),
+ (gst_install_plugins_installation_in_progress):
+ * gst-libs/gst/utils/install-plugins.h:
+ API: add API for applications to initiate installation of missing
+ plugins, ie. gst_install_plugins_async() primarily.
+ Based on libgimme-codec by Ryan Lortie.
+ * configure.ac:
+ Add --with-install-plugins-helper configure option so distros can specify
+ the path of the helper script or program to call when plugin installation
+ is requested (distros: please do any argument munging in this helper
+ script instead of patching GStreamer to pass arguments differently
+ to another program directly).
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Build and document new API.
+ * tests/check/libs/utils.c: (result_cb),
+ (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
+ (libgstbaseutils_suite):
+ Some simple checks for the new API.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads):
- Handle EOS properly.
+2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+ tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (test_float_conversion):
+ Add small test for 32bit float <=> 64bit float conversion (works
+ only one way so far, 32=>64 produces structured noise).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_sinkconnect),
- (gst_faad_chain), (gst_faad_change_state):
- * ext/faad/gstfaad.h:
- Allow playback of raw (unframed) MPEG AAC files (#148993).
+ gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (set_structure_widths_32_and_64), (make_lossless_changes):
+ We don't support floats with a width of 40, 48 or 56 bits.
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float), (double),
+ (audio_convert_get_func_index):
+ * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
+ (make_lossless_changes):
+ Support for 64-bit float audio in audioconvert (#339837)
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Throw error if we didn't recognize the stream. Fixes #152289.
+2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/: Add German translation (#352069).
+ Original commit message from CVS:
+ Patch by: Holger Wansing <linux wansing-online de>
+ * po/LINGUAS:
+ * po/de.po:
+ Add German translation (#352069).
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- Fix negotiation.
+2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-10-01 Francis Labonte <francis_labonte@hotmail.com>
+ ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
+ Original commit message from CVS:
+ reviewed by: Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
+ (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
+ Use newly added GstCollectPads API to free the allocated resources in
+ the GstOggPad structures (#402393).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak.
+ gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_vis_element):
+ Add audioresample+audioconvert in front of the visualisation
+ element, so that elements like libvisual 0.4 that don't support all
+ samplerates can work.
+ Fixes: #402505
-2004-10-01 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
+ (gst_play_base_bin_get_streaminfo_value_array):
+ Take some locks and make a copy of the streaminfo value array we
+ maintain while holding the lock, so that the application can
+ retrieve the stream-info as a value array in a thread-safe way.
- * sys/sunaudio/gstsunaudio.c: (gst_sunaudiosink_setparams):
- Solve #152805.
- * sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_mute):
- Solve 152806.
+2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ Don't fail on 0 sized buffers. Fixes #396835.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_audio_caps_with_data):
- Add codec_data handling (like asfdemux used to do).
- * gst/asfdemux/gstasf.c: (plugin_init):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
- Use riff-media for caps creation instead of our own (mostly
- broken) copy of its functions.
-
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_try_capture):
- Don't actually error out if we get another return value than
- -EINVAL. Opposite to what I first thought, drivers have random
- return values for this, although -EINVAL is the expected return
- value. Since this is not fatal, we shouldn't use
- GST_ELEMENT_ERROR() but just GST_ERROR_OBJECT().
-
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
- (dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
- (dvdreadsrc_get_property), (_open), (_seek), (_read),
- (dvdreadsrc_get), (dvdreadsrc_open_file),
- (dvdreadsrc_change_state):
- Fix. Don't do one big huge loop around the whole DVD, that will
- cache all data and thus eat sizeof(dvd) (several GB) before we
- see something.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
- Actually NULL'ify event after using it.
- * gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
- (gst_ebml_read_handle_event), (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_skip):
- Handle events.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
- (gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
- (gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
- Fix timing (this will probably break if I seek using menus, but
- I didn't get there yet). VOBs and normal DVDs should now work.
- Add a mpeg2-only pad with high rank so this get autoplugged for
- MPEG-2 movies.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
- (gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
- (gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
- (gst_mpeg_demux_get_audio_stream),
- (gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
- (gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
- Use this as second rank for MPEG-1 and MPEG-2. Still use this for
- MPEG-1 but use dvddemux for MPEG-2.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
- (gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
- (gst_mpeg_parse_parse_packhead):
- Timing. Only add pad template if it exists. Add sink template from
- class and not from ourselves. This means we will always use the
- correct sink template even if it is not the one defined in this
- file.
-
-2004-09-29 Wim Taymans <wim@fluendo.com>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_packet),
- (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
- Fix playback of mpeg again, timestamps where screwed up by
- patch 1.61.
-
-2004-09-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_src_query):
- Only return true if we actually filled something in. Prevents
- player applications from showing a random length for flac files.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
- (gst_riff_read_use_event), (gst_riff_read_handle_event),
- (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
- (gst_riff_read_strf_vids_with_data),
- (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
- OK, ok, so I implemented event handling. Apparently it's normal
- that we receive random events at random points without asking
- for it.
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
- (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
- (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
- (gst_avi_demux_sync), (gst_avi_demux_stream_scan),
- (gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
- (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
- (gst_avi_demux_stream_data), (gst_avi_demux_loop):
- * gst/avi/gstavidemux.h:
- Implement non-lineair chunk handling and subchunk processing.
- The first solves playback of AVI files where the audio and video
- data of individual buffers that we read are not synchronized.
- This should not happen according to the wonderful AVI specs, but
- of course it does happen in reality. It is also a prerequisite for
- the second. Subchunk processing allows us to cut chunks in small
- pieces and process each of these pieces separately. This is
- required because I've seen several AVI files with incredibly large
- audio chunks, even some files with only one audio chunk for the
- whole file. This allows for proper playback including seeking.
- This patch is supposed to fix all AVI A/V sync issues.
- * gst/flx/gstflxdec.c: (gst_flxdec_class_init),
- (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
- Work.
- * gst/modplug/gstmodplug.cc:
- Proper return value setting for the query() function.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Being in non-playing state (after, e.g., EOS) is not necessarily
- a bad thing. Allow for that. This fixes playback of short files.
- They don't actually playback fully now, because the clock already
- runs. This means that small files (<500kB) with a small length
- (<2sec) will still not or barely play. Other files, such as mod
- or flx, will work correctly, however.
-
-2004-09-28 Wim Taymans <wim@fluendo.com>
-
- * ext/speex/gstspeex.c: (plugin_init):
- * ext/speex/gstspeexdec.c: (gst_speex_dec_base_init),
- (gst_speex_dec_class_init), (speex_dec_get_formats),
- (speex_get_event_masks), (speex_get_query_types),
- (gst_speex_dec_init), (speex_dec_convert), (speex_dec_src_query),
- (speex_dec_src_event), (speex_dec_event), (speex_dec_chain),
- (gst_speexdec_get_property), (gst_speexdec_set_property),
- (speex_dec_change_state):
- * ext/speex/gstspeexdec.h:
- * ext/speex/gstspeexenc.c: (gst_speexenc_get_formats),
- (gst_speexenc_get_type), (speex_caps_factory), (raw_caps_factory),
- (gst_speexenc_base_init), (gst_speexenc_class_init),
- (gst_speexenc_sinkconnect), (gst_speexenc_convert_src),
- (gst_speexenc_convert_sink), (gst_speexenc_get_query_types),
- (gst_speexenc_src_query), (gst_speexenc_init),
- (gst_speexenc_get_tag_value), (comment_init), (comment_add),
- (gst_speexenc_metadata_set1), (gst_speexenc_set_metadata),
- (gst_speexenc_setup), (gst_speexenc_buffer_from_data),
- (gst_speexenc_push_buffer), (gst_speexenc_set_header_on_caps),
- (gst_speexenc_chain), (gst_speexenc_get_property),
- (gst_speexenc_set_property), (gst_speexenc_change_state):
- * ext/speex/gstspeexenc.h:
- Rewrote speex encoder, make sure it can be embedded in ogg.
- Implemented speex decoder.
-
-2004-09-28 Christian Schaller <christian@fluendo.com>
+2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
- * configure.ac:
- Remove kioslave plugin. Markey is brewing a new working one
- * ext/Makefile.am: Remove kioslave plugin
- * ext/kio: remove
- * gst-plugins.spec.in: remove kio plugin from spec
-
-2004-09-27 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
- (is_sync_frame), (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Make syncing to keyframes actually work for new clients and lagging
- clients.
+ gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ Detect BBCD as video/x-dirac, so we can play raw dirac
+ streams.
-2004-09-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/debug/gstnavigationtest.c: (gst_navigationtest_class_init),
- (gst_navigationtest_handle_src_event), (draw_box_planar411),
- (gst_navigationtest_planar411), (gst_navigationtest_change_state):
- * gst/debug/gstnavigationtest.h:
- make navigationtest display button-press and button-release events
+ ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
+ Original commit message from CVS:
+ * ext/theora/theoraenc.c: (theora_enc_chain):
+ Check return value of theora_encode_header(), or we might try to
+ allocate a random number of bytes. theora_encode_header() can fail
+ if libtheora has been compiled with encoding support disabled.
+ Fixes #398110.
-2004-09-26 Iain <iaingnome@gmail.com>
+2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/gst/.cvsignore: Do as buildbot says.
+ Original commit message from CVS:
+ * tests/check/gst/.cvsignore:
+ Do as buildbot says.
+
+2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_src_setcaps):
+ Fix strides in libvisual. Gst uses X strides.
+ Inspired by: <ed at catmur dot co dot uk> and
+ <tim at centricular dot net>
+ Fixes #401118.
+
+2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
+ (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
+ (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
+ (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_bisect_forward_serialno),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
+ (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
+ (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
+ (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
+ * ext/ogg/gstoggdemux.h:
+ Properly propagate streaming errors when we are scanning the file for
+ chains so that we don't crash when shut down. Might fix some crashers
+ when quickly switching oggs in RB such as #332503 and #378436.
+
+2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
+ Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
+ error code as well.
- * gst/interleave/interleave.c (all_channels_new_media): Checks if all
- the channels have received a new media event.
- (interleave_buffered_loop): Compresses a new media event on all
- channels into one.
+2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-09-26 Iain <iaingnome@gmail.com>
+ gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (remove_source):
+ Don't try to disconnect a signal from a finalized object.
- * gst/wavenc/gstwavenc.c (gst_wavenc_chain): Company says we need to
- call the sinkpad's default event handler and not the srcpads. He also
- says this is confusing :)
- (gst_wavenc_stop_file): Company says that seek events only go upstream
- we should send a discontinuous downstream instead.
+2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-25 Christian Schaller <christian@fluendo.com>
+ gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
+ Cast lock macro parameters to make sure we're actually accessing the
+ lock member at the right class level. Free list itself in _dispose()
+ as well and NULL it in case dispose gets called multiple times.
- * Update SPEC file to be usable in conjunction with Fedora Core,
- Fedora.us and freshrpms packages
- * Fix typo in multifilesrc test Makefile
+2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-09-24 Wim Taymans <wim@fluendo.com>
+ gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ (gst_decode_bin_dispose),(gst_decode_bin_finalize):
+ Free GstDecodeGroups no longer used.
+ (gst_decode_group_expose):
+ Don't unlock too many times !
+ (deactivate_free_recursive):
+ Free iterator once we're done with it.
+ Fix for recursively deactivating elements (stop at ghostpads).
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Only signal the no_more_pads signal when we have
- added the stream to our list.
+2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-24 Wim Taymans <wim@fluendo.com>
+ gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (handoff):
+ Fix up caps on the frame buffer before we save it and potentially
+ make it accessible to other threads via g_object_get; also use
+ gst_buffer_replace() instead of gst_mini_object_replace().
- * gst/playback/gstplaybasebin.c: (remove_prerolls),
- (new_decoded_pad):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (setup_sinks):
- Don't try to preroll or decode more than one audio/video
- track.
-
-2004-09-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Throw error if we failed to find a suitable output. This should
- throw an error if we successfully set up a pipeline (e.g. because
- we recognized a media file) but found no decodable streams in it
- (e.g. because it contains only media stream types for which we
- have no decoders, or because it's not a media type).
-
-2004-09-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/dirac/Makefile.am:
- * ext/dirac/gstdirac.cc:
- * ext/dirac/gstdiracdec.cc:
- * ext/dirac/gstdiracdec.h:
- Do something. Don't actually know if this works because I don't
- have a demuxer yet.
- * ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
- Add channels=1 to caps returned from _getcaps().
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
- (gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
- (gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
- (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
- (gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
- (gst_ogm_parse_change_state):
- Separate between audio/video so ogmaudioparse actually uses the
- audio pad templates. Both audio and video work now, including
- autoplugging. Also use sometimes-srcpad hack.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
- Handle events better. Don't hang on infinite loops.
- * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
- (gst_avi_demux_init), (gst_avi_demux_reset),
- (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
- (gst_avi_demux_change_state):
- * gst/avi/gstavidemux.h:
- Improve A/V sync. Still not perfect.
- * gst/matroska/ebml-read.c: (gst_ebml_read_seek),
- (gst_ebml_read_skip):
- Handle events better.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (gst_qtdemux_loop_header), (qtdemux_parse_trak),
- (qtdemux_audio_caps):
- Add IMA4. Improve event handling. Save offset after a seek when
- the headers are at the end of the file so that we don't end up in
- an infinite loop.
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- Add low-priority typefind support for files with no length.
-
-2004-09-23 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * testsuite/multifilesink/Makefile.am:
- fix typo
-
-2004-09-22 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls): Fix
- mistakes from thaytan's patches.
-
-2004-09-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
- For completeness, XSync in the destroy function as xvimage does.
-
-2004-09-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
- Correct caps negotiation
- * gst/volume/gstvolume.c: (volume_chain_float),
- (volume_chain_int16):
- Modify debug output to be little more informative
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_destroy):
- Add XSync calls after detaching from the shared memory segment to
- avoid a crash.
-
-2004-09-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_loop):
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_chain):
- * ext/vorbis/vorbisenc.h:
- remove explicit newmedia support from oggmux and vorbisenc
- add debug category to vorbisenc
- * gst/multifilesink/gstmultifilesink.c:
- (gst_multifilesink_class_init), (gst_multifilesink_init),
- (gst_multifilesink_dispose), (gst_multifilesink_set_location),
- (gst_multifilesink_set_property), (gst_multifilesink_next_file),
- (gst_multifilesink_handle_event), (gst_multifilesink_chain),
- (plugin_init):
- * gst/multifilesink/gstmultifilesink.h:
- add support for streamheader in multifilesink
-
-2004-09-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
- (gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
- (gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
- Prevent infinite loops. More correct error reporting.
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Error out if negotiation fails.
- * gst/playback/gstplaybasebin.c: (setup_source),
- (gst_play_base_bin_change_state), (gst_play_base_bin_error),
- (gst_play_base_bin_found_tag):
- Error/tag forwarding. Pre-roll fixes for source errors on state
- changes (e.g. "file does not exist") to prevent hangs.
-
-2004-09-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * testsuite/multifilesink/Makefile.am:
- * testsuite/multifilesink/lame_test.c: (gst_newmedia_base_init),
- (gst_newmedia_class_init), (gst_newmedia_init),
- (gst_newmedia_chain), (gst_newmedia_trigger), (test_format),
- (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/multifilesrc_test.c: (main):
- * testsuite/multifilesink/oggtheora_test.c:
- (gst_newmedia_base_init), (gst_newmedia_class_init),
- (gst_newmedia_init), (gst_newmedia_chain), (gst_newmedia_trigger),
- (test_format), (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/oggvorbis_test.c:
- (gst_newmedia_base_init), (gst_newmedia_class_init),
- (gst_newmedia_init), (gst_newmedia_chain), (gst_newmedia_trigger),
- (test_format), (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/wavenc_test.c: (gst_newmedia_base_init),
- (gst_newmedia_class_init), (gst_newmedia_init),
- (gst_newmedia_chain), (gst_newmedia_trigger), (test_format),
- (newfile_signal), (test_signal), (main):
- New media tests
-
-2004-09-20 Christian Schaller <christian@fluendo.com>
-
- * Fix mikmod license to LGPL as they have relicensed
- * Move Dirac and Effectv into LGPL section of README_license
-
-2004-09-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset),
- (gst_mad_change_state):
- Allow for mp3 rate/channels changes. However, only very
- conservatively. Reason that we *have* to enable this is smiply
- because the mad find_sync() function is not good enough, it will
- regularly sync on random data as valid frames and therefore make
- us provide random caps as *final* caps of the stream. The best fix
- I could think of is to simply require several of the same stream
- changes in a row before we change caps.
- The actual testcase that works now is #
- * ext/ogg/Makefile.am:
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstogmparse.c:
- OGM support (video only for now; I need an audio sample file).
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
- (gst_asf_demux_add_video_stream):
- WMV extradata.
- * gst/playback/gstplaybasebin.c: (unknown_type):
- Don't error out on single unknown-types after all. It's wrong.
- If we found type of video and audio but not of a subtitle stream,
- it will still error out (which is unwanted). Will find a better fix
- later on.
- * gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
- (ogmaudio_type_find), (plugin_init):
- OGM support.
-
-2004-09-20 Johan Dahlin <johan@gnome.org>
-
- * ext/jpeg/gstjpegdec.c (gst_jpegdec_chain): Allocate the buffer
- after setting caps.
-
-2004-09-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst/wavenc/gstwavenc.c: (gst_wavenc_init), (gst_wavenc_chain):
- * gst/wavenc/gstwavenc.h:
- Added newmedia support to wavenc
-
-2004-09-17 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstfdset.c: (gst_fdset_fd_has_closed),
- (gst_fdset_fd_has_error), (gst_fdset_fd_can_read),
- (gst_fdset_fd_can_write), (gst_fdset_wait):
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_get_stats),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_handle_clients),
- (gst_multifdsink_close), (gst_multifdsink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_removed):
- Small cleanups in fdset.c
- Use a hastable to map fd to the client structure for faster
- lookup in _remove and get_stats.
- Added virtual function to close the fds.
- Handle clients even when the select/poll call was unblocked because
- of a command.
- Implement syncing to keyframe in the recovery procedure.
+2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-16 Iain <iaingnome@gmail.com>
+ gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
+ Make getting the current frame thread-safe.
- * gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int): Free the
- try caps.
+2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
+ (gst_decode_group_new), (gst_decode_group_free):
+ Set queues to bigger sizes to cope with HD contents.
+ Fix some mutex freeing and add comment about MT safe methods.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_get_audio_stream):
- Caps are only set if the type of the stream is unknown, but this
- is initialized in ->init_stream(), so set to UNKNOWN after calling
- ->init_stream() so that capsnego starts.
+2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
+ (gst_text_overlay_text_event):
+ Don't unnecessarily ref (and then leak) upstream events if the text
+ pad is not linked. Fixes #399948.
+ * tests/check/gst-plugins-base.supp:
+ Add suppression for pango on edgy/x86 for textoverlay test.
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_data):
- Just hardcode for raw audio then. AVI audio sucks.
+2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-09-15 Arwed v. Merkatz <v.merkatz@gmx.net>
+ gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Add some more fixed payloads.
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
- * gst/matroska/matroska-mux.c: (audiosink_templ),
- (gst_matroska_mux_audio_pad_link):
- * gst/typefind/gsttypefindfunctions.c: (tta_caps), (plugin_init):
- Use audio/x-ttafile for tta files and audio/x-tta for raw tta frames.
+2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
+ Error out properly if we get an error from libogg while reading the
+ BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_data):
- Try to fix a/v sync issues.
+2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-15 David Schleef <ds@schleef.org>
+ gst/playback/gstdecodebin2.c: Don't leak mutex.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
+ Don't leak mutex.
+ * tests/check/elements/playbin.c:
+ (test_sink_usage_video_only_stream),
+ (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
+ (test_suburi_error_wrongproto), (test_missing_urisource_handler),
+ (test_missing_suburisource_handler),
+ (test_missing_primary_decoder), (playbin_suite):
+ Run all tests once with decodebin and once with decodebin2.
+ One test does not pass yet with decodebin2.
- * configure.ac: remove NASM check, since we don't use it. Update
- dirac check to 0.4
- * ext/dirac/gstdiracdec.cc: update to current 0.4 API
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Initialized variables.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
- (gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
- SVQ3 format
+2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
+ Fix the cases where oggmux doesn't properly figure out that all
+ sinkpads have gone EOS, and therefore doesn't push out the remaining
+ buffers and the final EOS event.
+ Fixes #363379
+
+2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Don't lock on navigation event push, just on keysym to string.
+ Original commit message from CVS:
+ 2007-01-23 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
+ Don't lock on navigation event push, just on keysym to string.
+ Fixes #397673 again.
+
+2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin2.c: Cleanups.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
+ (get_current_group), (group_demuxer_event_probe),
+ (gst_decode_group_expose), (deactivate_free_recursive),
+ (gst_decode_group_free):
+ Cleanups.
+ Don't forget to emit 'no-more-pads' once a group is exposed.
+ Cleanup elements from a DecodeGroup once we remove it.
+ Protect call to gst_decode_group_expose() with the decodebin lock.
+
+2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
+ Original commit message from CVS:
+ 2007-01-22 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
+ Looking at Xorg code i can't figure out if that XKeysymToString
+ function is thread sensible or not. Lock it just in case as
+ recommended by Radek Doulik <rodo at ximian dot com>.
+
+2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Lock that X Call as well. Fixes #397673.
+ Original commit message from CVS:
+ 2007-01-22 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
+ Lock that X Call as well. Fixes #397673.
+
+2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
+ Don't go into an endless loop if the file starts with 00 00 01 2X,
+ like quicktime redirect files might. Fixes #396042.
+ * tests/check/Makefile.am:
+ * tests/check/gst/.cvsignore:
+ * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
+ (typefindfunctions_suite):
+ Add unit test for the above.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_data):
- * gst/avi/gstavidemux.h:
- Fix for compressed audio (mp3) timestamp generation. How did this
- ever work?
+ gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ On second thought, use "depth" field rather than "bpp" field.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Volume is a double not a float.
+ gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Camtasia caps apparently need a bpp field (#398875).
-2004-09-15 Wim Taymans <wim@fluendo.com>
+2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_remove_client_link),
- (gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
- Don't close the fd in multifdsink as we didn't open it in the
- first place. Some cleanups.
+ gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_subtitle),
+ (gen_source_element), (gst_play_base_bin_change_state):
+ Attempt at a better error message in case we don't have the required
+ URI handler installed; post missing-plugin message also when we're
+ missing an URI handler for the subtitle URI; clean up properly also
+ when an error occurs and we never made it to PAUSED state.
+ * tests/check/elements/playbin.c: (GST_START_TEST),
+ (playbin_suite):
+ Check that we're also getting a missing-plugin messsage for a
+ missing subtitle URI handler (and clean up properly).
-2004-09-15 Wim Taymans <wim@fluendo.com>
+2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- Fix the case where the muxer would mark pages as delta
- frames when they are not (vorbis only ogg).
+ gst/playback/gstplaybasebin.c: Plug a few reference leaks.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
+ Plug a few reference leaks.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (state_change), (setup_source),
- (gst_play_base_bin_change_state):
- Handle the case where we failed to setup a clear pipeline. This
- will throw an error (or EOS, another nice case) and if you don't
- catch that, the app will wait for the signal forever (and thus
- hang).
+ gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
+ Lower probability a bit if the marker isn't right at the start,
+ to decrease the chance of false positives.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/gnomevfs/gstgnomevfssink.c:
- (gst_gnomevfssink_uri_get_protocols):
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnomevfssrc_uri_get_protocols):
- * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
- * ext/gnomevfs/gstgnomevfsuri.h:
- Use _uri_new() instead of _open(), so it doesn't take as long and
- Christophe's computer won't hang.
- * gst/playback/gstplaybasebin.c: (unknown_type):
- Throw error on unknown media type, so apps actually display it.
+ gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
+ Small mpeg2 system stream typefinding improvement: make typefinder
+ probe a bit into the stream instead of just looking for a marker
+ at the beginning. Fixes #397810.
-2004-09-14 Brian Cameron <brian.cameron@sun.com
+2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * tools/gst-launch-ext-m.m: Changed ~ to $ENV{HOME} to allow
- this script to work on Solaris since bash shell handles echo
- differenly than bash.
+ gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c:
+ Remove compatibility cruft for prehistoric GLib versions.
-2004-09-17 Wim Taymans <wim@fluendo.com>
+2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (queue_overrun), (no_more_pads),
- (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_add_element):
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event):
- Some more work on making sure seeking pauses the pipeline and
- that changing the uri actually does something.
+ gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad):
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_handle_message_func), (unknown_type):
+ Let decodebin be the element to post missing-plugin messages for
+ missing decoders (rather than playbin); make playbin implement
+ GstBin::handle_message so we can suppress missing-plugin messages
+ for types we're not handling on purpose (don't want to bring up an
+ installer in those cases).
-2004-09-17 Wim Taymans <wim@fluendo.com>
+2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstfdset.c: (gst_fdset_wait):
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_close):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send),
- (gst_tcpserversink_close):
- Be a bit more paranoid when freeing memory.
+ gst/: Fix potentially unaligned access (#397207).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ * gst-libs/gst/tag/gstvorbistag.c:
+ (gst_tag_list_to_vorbiscomment_buffer):
+ * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
+ Fix potentially unaligned access (#397207).
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
- (qtdemux_parse_trak):
- Don't crash by dividing by zero (see sample movie in #126922).
+ tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (set_scale), (update_scale),
+ (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
+ (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
+ (main):
+ Allow to toggle looping while it plays. Fix callback prototype. Clean
+ up code a bit more. Add copyright header.
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Don't touch non-existing data (fixes crash on file in #140147).
+ sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ Red and blue mask was swapped (spotted by Dan Williams).
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstplaybasebin.c:
- (gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
- Handle double disposals, and proper change of URIs.
+ gst-libs/gst/tag/: Use new beats-per-minute tag from core.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Use new beats-per-minute tag from core.
-2004-09-13 Martin Eikermann <meiker@upb.de>
+2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegparse.c:
- fix synchronistation for streams recorded from digital PCR
- fixes bug #119376
+ po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Add new files with translatable strings, so they actually make it
+ into the template file one day.
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_get_type),
- (gst_gnomevfssink_dispose), (gst_gnomevfssink_init),
- (gst_gnomevfssink_uri_get_type),
- (gst_gnomevfssink_uri_get_protocols),
- (gst_gnomevfssink_uri_get_uri), (gst_gnomevfssink_uri_set_uri),
- (gst_gnomevfssink_uri_handler_init),
- (gst_gnomevfssink_set_property), (gst_gnomevfssink_get_property),
- (gst_gnomevfssink_open_file), (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
- (gst_gnomevfssrc_init), (gst_gnomevfssrc_dispose),
- (gst_gnomevfssrc_uri_get_type),
- (gst_gnomevfssrc_uri_get_protocols), (gst_gnomevfssrc_uri_get_uri),
- (gst_gnomevfssrc_uri_set_uri), (gst_gnomevfssrc_uri_handler_init),
- (gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
- (gst_gnomevfssrc_open_file), (gst_gnomevfssrc_close_file):
- * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
- * ext/gnomevfs/gstgnomevfsuri.h:
- Add URI support to Gnome-VFS plugins. Tries to load a fixed list
- of fake URIs to see which this version of Gnome-VFS likes, and
- uses that for the Gst-URI interface. Makes playbin support http://
- streams. Also fix up some stupid behaviour in gnomevfssrc.
-
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_update),
- (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
- (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
- (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
- Update mixer (to sync with other sessions) if we try to obtain
- a new value. This makes alsamixer work accross applications.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
- Only call sync functions if we're running, else alsalib asserts.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
- Sometimes fails to compile. Possibly a gcc bug.
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- Add a reference to an application-provided object, because we lose
- this same reference if we add it to the bin. If we don't do this,
- we can only use this object once and thus crash if we go from
- ready to playing, back to ready and back to playing again.
- Also add an audioscale element because several cheap soundcards -
- like mine - don't support all samplerates.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_xcontext_clear), (gst_ximagesink_change_state):
- Fix wrong order or PAR calls. Makes automatically obtained PAR
- from the X server atually being used.
-
-2004-09-12 David Schleef <ds@schleef.org>
-
- Fixes: #151879, #151881, #151882, #151883, #151884, #151886,
- #151887, #152102, #152247.
- * examples/indexing/indexmpeg.c: 64-bit warning fixes.
- * examples/seeking/cdparanoia.c: same
- * examples/seeking/cdplayer.c: same
- * examples/seeking/seek.c: same
- * examples/seeking/spider_seek.c: same
- * examples/seeking/vorbisfile.c: same
- * examples/stats/mp2ogg.c: same
- * ext/esd/esdsink.c: (gst_esdsink_class_init),
- (gst_esdsink_dispose): Dispose of element properly.
- * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_seek): 64-bit warning
- fixes.
- * ext/nas/nassink.c: (gst_nassink_class_init),
- (gst_nassink_dispose): Dispose of element correctly.
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Fix leak.
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
- Fix 64-bit warning.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy):
- Fix 64-bit warning.
-
-2004-09-12 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * configure.ac : change speex detection as 1.1.6 now uses
- .pc/pkg-config and they changed their headers location.
-
-2004-09-09 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * gst/matroska/matroska-mux.h:
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
- (gst_matroska_mux_start), (gst_matroska_mux_finish),
- (gst_matroska_mux_write_data):
- Write multiple blocks/frames per cluster.
- Write meta-seek information (seek heads).
-
-2004-09-09 Scott Wheeler <wheeler@kde.org>
-
- * gst/gstplaybin.c: (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (gen_audio_element), (gen_audio_element):
- Add a volume element / property to the pipeline.
-
-2004-09-07 Wim Taymans <wim@fluendo.com>
-
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_buffers):
- Copy timestamps from the master pad to the output buffers.
-
-2004-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/raw1394/gstdv1394src.c:
- throw errors when applicable
-
-2004-09-01 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * gst/matroska/ebml-ids.h:
- * gst/matroska/ebml-read.c: (gst_ebml_read_date):
- * gst/matroska/ebml-write.c: (gst_ebml_write_date):
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_finish):
- automatically convert unix time <-> ebml time when reading/writing
- a date, use gst_ebml_write_uint to write CUETIME,
- not gst_ebml_write_date.
- * gst/matroska/matroska-ids.h:
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_create_uid),
- (gst_matroska_mux_reset), (gst_matroska_mux_audio_pad_link),
- (gst_matroska_mux_track_header), (gst_matroska_mux_start),
- (gst_matroska_mux_write_data):
- Write track and segment UIDs, write muxing date, write
- TRACKDEFAULTDURATION for TTA audio, write BLOCKDURATION if known.
- Create cues for audio only files.
-
-2004-08-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Re-commit ALSA switches.
- * gst/adder/gstadder.c: (gst_adder_loop):
- 64-bit fix (#151416).
- * gst/debug/progressreport.c: (gst_progressreport_report):
- 64-bit fix (#151419).
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_contents):
- 64-bit fix (#151420).
- * gst/playback/test3.c: (update_scale):
- 64-bit fix (#151421).
-
-2004-08-31 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ 2007-01-12 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
+ (gst_base_audio_sink_activate_pull): Remove the handwavey nego
+ stuff, as the base class handles this now. Actually tell the ring
+ buffer to start.
+ (gst_base_audio_sink_callback): Cast the ring buffer correctly.
+ How did this work before? Maybe I'm not as awesome a programmer as
+ I think.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c
+ (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
+ of a pad function.
+
+2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
+ Original commit message from CVS:
+ * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
+ Remove more fields so that the application can better blacklist
+ formats that have been tried before.
- * configure.ac:
- bump nano to cvs
+2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
-=== release 0.8.4 ===
+ * gst-plugins-base.spec.in:
+ add latest files
+ Original commit message from CVS:
+ add latest files
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: releasing 0.8.4, "Alias"
+ gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/mixerutils.h:
+ Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
+ used when compiling with c++ compilers as well.
-2004-08-31 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/Makefile.am:
- fix makefile. Fixes #151462.
+ gst/typefind/gsttypefindfunctions.c: Fix comment.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ Fix comment.
-2004-08-30 Wim Taymans <wim@fluendo.com>
+2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstfdset.c: (gst_fdset_free), (gst_fdset_wait):
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write):
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init_send):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init_receive):
- Fix some memory leaks.
+ gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (post_missing_element_message),
+ (gen_video_element), (gen_text_element), (gen_audio_element),
+ (gen_vis_element):
+ Post missing-plugin messages also when we error out because
+ converters, textoverlay or auto*sinks are missing (#161922).
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
+ (is_demuxer_element), (new_caps):
+ * gst/playback/gstplaybasebin.c: (source_new_pad):
+ Fix the case where we try to ref a NULL element when we delay a link
+ because of unfixed caps.
+ Set the state of autoplugged decodebins to PAUSED.
+ RTSP now works in playbin, we can remove it from the blacklist.
+
+2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplaybasebin.c: (string_arr_has_str),
+ (unknown_type), (setup_subtitle), (gen_source_element):
+ * gst/playback/gstplaybin.c: (plugin_init):
+ Post missing-plugin messages on the bus for missing sources and
+ missing decoders/demuxers/depayloaders; fix error code used when
+ we're missing an URI handler source; for media types that we are not
+ handling on purpose at the moment, don't print "don't know how to
+ handle xyz" messages to the terminal or post missing-plugin
+ messages on the bus.
+ * tests/check/elements/playbin.c: (create_playbin),
+ (GST_START_TEST), (gst_codec_src_uri_get_type),
+ (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
+ (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
+ (gst_codec_src_init_type), (gst_codec_src_base_init),
+ (gst_codec_src_create), (gst_codec_src_class_init),
+ (gst_codec_src_init), (plugin_init), (playbin_suite):
+ Add some tests for the missing-plugin stuff.
- Patch by: David Schleef
+2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * sys/Makefile.am:
- rename our detection macro for V4L2. Fixes #151236.
+ API: add new libgstbaseutils library with functions
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/utils/Makefile.am:
+ * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
+ * gst-libs/gst/utils/base-utils.h:
+ * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
+ (find_format_info), (caps_are_rtp_caps),
+ (gst_base_utils_get_source_description),
+ (gst_base_utils_get_sink_description),
+ (gst_base_utils_get_decoder_description),
+ (gst_base_utils_get_encoder_description),
+ (gst_base_utils_get_element_description),
+ (gst_base_utils_add_codec_description_to_tag_list),
+ (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
+ * gst-libs/gst/utils/descriptions.h:
+ * gst-libs/gst/utils/missing-plugins.c:
+ (missing_structure_get_type), (copy_and_clean_caps),
+ (gst_missing_uri_source_message_new),
+ (gst_missing_uri_sink_message_new),
+ (gst_missing_element_message_new),
+ (gst_missing_decoder_message_new),
+ (gst_missing_encoder_message_new),
+ (missing_structure_get_string_detail),
+ (missing_structure_get_caps_detail),
+ (gst_missing_plugin_message_get_installer_detail),
+ (gst_missing_plugin_message_get_description),
+ (gst_is_missing_plugin_message):
+ * gst-libs/gst/utils/missing-plugins.h:
+ API: add new libgstbaseutils library with functions
+ - to create and parse missing-plugins messages
+ - that provide (translated) descriptions for caps/decoders/sources/etc.
+ Closes #392393.
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ Add new lib.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Generate docs for new lib and API.
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/utils.c: (missing_msg_check_getters),
+ (GST_START_TEST), (libgstbaseutils_suite):
+ Add some basic unit tests.
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- Patch by: David Schleef
+ ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
+ Original commit message from CVS:
+ * ext/ogg/Makefile.am:
+ Dist gstoggdemux.h to fix 'make distcheck'.
+ * sys/v4l/Makefile.am:
+ Fix 'make distcheck' even more.
- * configure.ac:
- check to define LAMEPRESET. Fixes #151232.
+2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Added docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
+ (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
+ (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
+ (gst_ogg_demux_perform_seek):
+ * ext/ogg/gstoggdemux.h:
+ Added docs.
+ Add some more comments.
+ Small cleanups.
+
+2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Small documentation updates/fixes
+ Original commit message from CVS:
+ * ext/theora/theoradec.c:
+ * ext/vorbis/vorbisdec.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Small documentation updates/fixes
+
+2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
+ Original commit message from CVS:
+ * configure.ac:
+ Require core CVS HEAD for Andy's basesrc/sink API additions.
-2004-08-27 David Schleef <ds@schleef.org>
+2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
- * sys/glsink/glimagesink.c: (gst_glimagesink_ximage_put),
- (gst_glimagesink_xwindow_new), (gst_glimagesink_xcontext_get),
- (gst_glimagesink_fixate): Move local variable declarations to
- make gcc-2.95 happy.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
+ Original commit message from CVS:
+ Patch by: Günter Thelen <daedalus dot inc at gmx net>
+ * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
+ (plugin_init):
+ Add typefinder for flac-in-ogg in conformance with the ogg-mapping
+ on flac.sf.net (there appear to be other versions of the first
+ ogg page in the wild) (#391365).
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- bump nano for prerelease
+ configure.ac: Check if localtime_r() is available.
+ Original commit message from CVS:
+ * configure.ac:
+ Check if localtime_r() is available.
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
+ If localtime_r() is not available, fall back to localtime(). Should
+ fix build on MingW (#393310).
-2004-08-27 David Schleef <ds@schleef.org>
+2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/sunaudio/Makefile.am: Add sunaudiosrc patch from Bala
- * sys/sunaudio/gstsunaudiosrc.c:
- * sys/sunaudio/gstsunaudiosrc.h:
+ gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (parse_mdvdsub):
+ * gst/subparse/gstsubparse.h:
+ Remove spurious 1000 subtrahend when calculating the timestamp from
+ the frame number and the frame rate . Also, use the frames/second
+ value specified in the first line of the file, if one is specified
+ there. Should fix #357503.
+ * tests/check/elements/subparse.c: (do_test),
+ (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
+ (subparse_suite):
+ Add some basic unit tests for the microdvd subtitle format.
-2004-08-27 Arwed v. Merkatz <v.merkatz@gmx.net>
+2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ sys/xvimage/xvimagesink.c: Fixes : #390076.
+ Original commit message from CVS:
+ 2007-01-07 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_xvimage_put),
+ (gst_lookup_xv_port_from_adaptor),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_set_xwindow_id),
+ (gst_xvimagesink_set_event_handling),
+ (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
+ (gst_xvimagesink_init), (gst_xvimagesink_class_init):
+ Patch by : Young-Ho Cha <ganadist at chollian dot net>
+ Fixes : #390076.
+ Add an adaptor property to select a specific XV adaptor.
+ * sys/xvimage/xvimagesink.h:
+
+2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Use flow_lock much more to protect every access to xwindow.
+ Original commit message from CVS:
+ 2007-01-07 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
+ (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
+ (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
+ (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
+ (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
+ (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_change_state),
+ (gst_xvimagesink_set_xwindow_id),
+ (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
+ Use flow_lock much more to protect every access to xwindow.
+ Try to catch erros while creating images in case some drivers
+ are
+ just generating an XError when the requested image is too big.
+ Should fix : #354698, #384008, #384060.
+ * tests/icles/stress-xoverlay.c: (cycle_window),
+ (create_window):
+ Implement some stress testing of setting window xid.
+
+2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/common/libgsaudio.def: Add new exported function.
+ Original commit message from CVS:
+ * win32/common/libgsaudio.def:
+ Add new exported function.
+ * win32/common/libgstogg.dsp:
+ Add gstoggaviparse.c to the build.
+ * win32/common/libgstvideoscale.dsp:
+ Add vs_4tap.c to the build.
+ * win32/common/libgstvorbis.dsp:
+ Add vorbistag.c to the build.
- * gst/matroska/ebml-read.c: (gst_ebml_peed_id), (gst_ebml_read_element_id),
- handle EOS correctly
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_link):
- * gst/matroska/matroska-mux.h:
- added BITMAPINFOHEADER structure, mux video/x-divx and video/x-xvid in
- VFW compatibility mode
+2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
+ Original commit message from CVS:
+ 2007-01-06 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_class_init)
+ (gst_base_audio_sink_init):
+ (gst_base_audio_sink_activate_pull): Add an activate_pull function
+ to baseaudiosink, and tell basesink that we can work in pull mode.
+ This way the ring buffer thread drives the pipeline directly, if
+ pull mode is possible. There is some lingering nastiness regarding
+ capsnego, however.
+ (gst_base_audio_sink_callback): Implement the callback to pull
+ data. This interface is a bit light, though -- it should get a
+ GstFlowReturn return value at least.
+
+2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- patch by: Zaheer Abbas Merali
+ Printf format and missing argument fixes.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ * gst/playback/gstdecodebin2.c:
+ (gst_decode_group_check_if_blocked):
+ Printf format and missing argument fixes.
- * ext/ogg/gstoggmux.c:
- * ext/vorbis/vorbisenc.c:
- * ext/vorbis/vorbisenc.h:
- handle NEWMEDIA
-
-2004-08-26 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * gst/matroska/ebml-write.c: (gst_ebml_write_float),
- fix byte order reversion on little endian machines.
- * gst/matroska/matroska-mux.c: (audiosink_templ),
- (gst_matroska_mux_audio_pad_link):
- add TTA codec to the list of supported codecs.
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_init),
- (gst_matroska_mux_start), (gst_matroska_mux_finish),
- (gst_matroska_mux_write_data):
- * gst/matroska/matroska-mux.h:
- write segment duration correctly, write muxing app string, fixes bugs
- #140897 and #140898.
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_loop),
- wait for all pads to be negotiated before starting to mux.
-
-2004-08-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain):
- * ext/lame/gstlame.h:
- Added new media support to lame
-
-2004-08-25 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup),
- send vorbis headers at the beginning of a stream, fixes bug #141554.
- Interpret BLOCKDURATION and set buffer duration accordingly, fixes
- bug #148950.
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps),
- (gst_matroska_demux_plugin_init):
- * gst/matroska/matroska-ids.h:
- enable demuxing of TTA audio streams, fixes bug #148951.
- * gst/typefind/gsttypefindfunctions.c: (tta_type_find), (plugin_init),
- enable typefinding for TTA audio files, fixes bug #148711.
- * ext/xvid/gstxviddec.c: (gst_xviddec_chain),
- set XVID_LOWDELAY flag for decoding so xvid always returns an image,
- fixes playback of packed bitstream and xvid with bframes, bug #135407.
-
-2004-08-24 Sebastien Cote <sc5@hermes.usherb.ca>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_element_data), (gst_riff_read_seek),
- (gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
- #144616, patch reviewed by Ronald and committed by Christophe Fergeau
- <teuf@gnome.org>
-
-2004-08-23 Iain <iaingnome@gmail.com>
-
- * ext/mad/gstid3tag.c (gst_mad_id3_to_tag_list): Special case COMM
- tags. They appear to be handled differently to normal.
- (tag_list_to_id3_tag_foreach): Ditto.
-
-2004-08-22 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- Make sure we never send -1 granulepos.
-
-2004-08-20 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_loop):
- I will accept bitchslappings with non sharp objects.
-
-2004-08-20 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- Clean up the test for lame presets
+ ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
+ (gst_ogm_parse_change_state):
+ Activate pads before adding them to the element.
-2004-08-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/lame/Makefile.am:
- * ext/lame/gstlame.c: (gst_lame_class_init),
- (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
- Only enable lame presets if version of lame has presets in API
-
-2004-08-19 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_get):
- * gst/udp/gstudpsrc.h:
- Don't call gst_pad_push in a get function. Fixes #150449
-
-2004-08-18 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstfdset.c: (gst_fdset_free), (gst_fdset_set_mode),
- (gst_fdset_get_mode), (gst_fdset_add_fd), (gst_fdset_remove_fd),
- (gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- * gst/tcp/gstfdset.h:
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write):
- * gst/tcp/gstmultifdsink.h:
- Some extra checks in gstfdset.
- Only use send() when the fd is a socket. Don't try to
- read from write only fds.
+ tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
+ Original commit message from CVS:
+ * tests/examples/seek/scrubby.c: (main):
+ * tests/examples/seek/seek.c: (main):
+ Call g_thread_init() first thing in main() (see #391278).
-2004-08-18 Wim Taymans <wim@fluendo.com>
+2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstfdset.c: (gst_fdset_add_fd), (gst_fdset_remove_fd),
- (gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- Add more locking and bounds checking.
+ tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/netbuffer.c: (GST_START_TEST),
+ (netbuffer_suite):
+ Add test for GstNetBuffer + gst_buffer_copy(). Disabled
+ for the time being, since it's broken, see #393099.
-2004-08-18 Wim Taymans <wim@fluendo.com>
+2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_wait):
- Realloc test fdset in the lock and right before starting
- the poll call. Bump the limit to 4096.
+ tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Update to use GST_PLUGINS_BASE_CFLAGS as well.
-2004-08-17 David Schleef <ds@schleef.org>
+2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: Fix caps to handle full range
- of rates and channels. Make debugging less obnoxious.
+ configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
+ Original commit message from CVS:
+ * configure.ac:
+ split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
+ so that GST_BASE_CFLAGS can go inbetween them, making sure
+ we use uninstalled gst-libs headers
+ * docs/libs/Makefile.am:
+ * ext/alsa/Makefile.am:
+ * ext/cdparanoia/Makefile.am:
+ * ext/gnomevfs/Makefile.am:
+ * ext/libvisual/Makefile.am:
+ * ext/ogg/Makefile.am:
+ * ext/theora/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst/adder/Makefile.am:
+ * gst/audioconvert/Makefile.am:
+ * gst/audiorate/Makefile.am:
+ * gst/audioresample/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/tcp/Makefile.am:
+ * gst/videoscale/Makefile.am:
+ * gst/volume/Makefile.am:
+ * sys/ximage/Makefile.am:
+ * sys/xvimage/Makefile.am:
+ * tests/icles/Makefile.am:
+ adapt
- Patch from Balamurali Viswanathan implementing a mixer for
- Sun audio. (bug #144091):
- * sys/sunaudio/gstsunelement.c:
- * sys/sunaudio/gstsunelement.h:
- * sys/sunaudio/gstsunmixer.c:
- * sys/sunaudio/gstsunmixer.h:
+2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
-2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
+ Original commit message from CVS:
+ 2007-01-04 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/interfaces/xoverlay.c:
+ (gst_x_overlay_handle_events):
+ * gst-libs/gst/interfaces/xoverlay.h:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
+ (gst_ximagesink_set_xwindow_id),
+ (gst_ximagesink_set_event_handling),
+ (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
+ (gst_ximagesink_get_property), (gst_ximagesink_init),
+ (gst_ximagesink_class_init):
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
+ (gst_xvimagesink_set_xwindow_id),
+ (gst_xvimagesink_set_event_handling),
+ (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
+ (gst_xvimagesink_get_property), (gst_xvimagesink_init),
+ (gst_xvimagesink_class_init):
+ * sys/xvimage/xvimagesink.h:
+ * tests/icles/stress-xoverlay.c: (toggle_events),
+ (create_window):
+ Add a method to the XOverlay interface to allow disabling of
+ event handling in x[v]imagesink elements. This will let X events
+ propagate to parent windows which can be usefull in some cases.
+ Be carefull that the application is then responsible of pushing
+ navigation events and expose events to the video sink.
+ Fixes: #387138.
+
+2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
+ (fixes #392070).
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/gstaudioscale.h:
- made audioscale resample from any sample rate to any sample rate
+2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ Dist design docs.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/design/Makefile.am:
+ Dist design docs.
- * ext/libpng/gstpngdec.c:
- error out on unsupported types
+2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
+
+ docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
+ Original commit message from CVS:
+ 2006-12-27 Julien MOUTTE <julien@moutte.net>
+ * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
+ documentation
+ typo. Fixes: #390063.
+
+2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Plug a caps leak.
+ Original commit message from CVS:
+ 2006-12-27 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
+ caps leak.
+ * win32/common/config.h: Updated.
+
+2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
+ (setup_gdpdepay_streamheader):
+ * tests/check/elements/gdppay.c: (cleanup_gdppay),
+ (setup_gdppay_streamheader):
+ Fix the dp tests, but activating the pads for the streamheader tests
+ too and cleaning up conditionaly
-2004-08-17 Iain <iaingnome@gmail.com>
+2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
+ (img_get_alpha_info):
+ Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
+ other end of the word. Fixes: #387073.
+ Add some inconsequential branch hints in a couple of places.
+
+2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_caps_to_smpfmt):
+ The "signed" field in raw audio caps is of boolean type, trying to
+ extract the value with _get_int() will fail (fix to keep in sync with
+ the copy in gst-ffmpeg)
- * ext/flac/gstflacenc.c (gst_flacenc_update_quality): Only set the
- mid_side and loose_mid_side properties if its a stereo stream.
+2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/elements/: consistent pad (de)activation
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c: (cleanup_audioresample):
+ * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
+ * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
+ (cleanup_gdpdepay):
+ * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
+ * tests/check/elements/subparse.c: (teardown_subparse):
+ * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
+ * tests/check/elements/videorate.c: (cleanup_videorate):
+ * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
+ * tests/check/elements/volume.c: (cleanup_volume):
+ * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
+ (cleanup_vorbisdec):
+ * tests/check/elements/vorbistag.c: (setup_vorbistag),
+ (cleanup_vorbistag):
+ consistent pad (de)activation
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/theoradec.c: (gst_theora_dec_class_init),
- (theora_get_formats), (theora_dec_src_convert),
- (theora_dec_sink_convert), (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_event), (theora_dec_chain):
- Add a debug line.
+ gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Forgot to register the extensions.
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate),
- (gst_ogg_pad_push):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_push_page),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- Mark delta units in the muxer.
- Try to decode the packet after an out-of-sync error from
- libogg.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
+ (plugin_init):
+ Add typefinder for VIVO files (my christmas present to the 90s).
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_set_property), (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Added option to send a keyframe to clients as the first buffer.
- Make timeout property writable.
+ gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (type_found):
+ Special-case the text/plain media type: we only want to recognise it
+ as a 'raw' decoded media type if it comes from a demuxer or subtitle
+ parser, but not if the entire stream is of text/plain type. If the
+ entire stream is text/plain, we should just error out.
+ This fixes playback of audio files with lyrics in totem. Totem can't
+ distinguish between text files and subtitle files and passes any
+ .txt file with the same basename as the main file to playbin as
+ suburi, and playbin will then throw a 'subtitle found, but no video
+ stream' error, which isn't entirely helpful. See #380342.
+ Also, with this change we'll show a slightly more correct error
+ message in case totem passes a playlist file to us (although a
+ custom error message wording instead of the default text would
+ probably not be a bad idea either).
+ Same problem also needs to be fixed for playbin+decodebin2.
+ * tests/check/Makefile.am:
+ * tests/check/elements/decodebin.c: (src_handoff_cb),
+ (decodebin_new_decoded_pad_cb), (GST_START_TEST),
+ (decodebin_suite):
+ Add simple unit test for decodebin for the above.
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
- patch by: Wim Taymans
+ gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
+ Refuse to change state to READY when we failed to create any of the
+ required elements in our instance init function.
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstmultifdsink.c:
- fix index comparison, should include 0
+2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-16 Wim Taymans <wim@fluendo.com>
+ docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Small docs fixes/updates.
+ * gst-libs/gst/video/gstvideosink.h:
+ Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
+ from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
+ removed from the base sink API between 0.9.6 and 0.9.7).
+ API: add GST_VIDEO_SINK_CAST and use it for the height/width
+ accessor macros, so we don't do a runtime GObject type check every
+ time we use them.
- * gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_new),
- (gst_fdset_add_fd), (gst_fdset_remove_fd),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- copy when reallocing for poll so the select arguments don't get
- changed during the call
+2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-08-16 Wim Taymans <wim@fluendo.com>
+ add doap file
+ Original commit message from CVS:
+ * Makefile.am:
+ * gst-plugins-base.doap:
+ * gst-plugins-base.spec.in:
+ add doap file
- * ext/theora/theoraenc.c: (gst_border_mode_get_type),
- (gst_theora_enc_class_init), (theora_enc_sink_link),
- (theora_buffer_from_packet), (theora_enc_chain):
- Fix bug where buffers were not marked as keyframes
- correctly.
+2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
-2004-08-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Declare variables at the beginning of a block. Fixes #383195.
+ Original commit message from CVS:
+ Patch by: Jens Granseuer <jensgr at gmx net>
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer):
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
+ Declare variables at the beginning of a block. Fixes #383195.
- * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type),
- (gst_lame_preset_get_type), (gst_lame_class_init):
- describe the enum values for vbr mode and presets more verbosely
+2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ configure.ac: Bump version nano - back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version nano - back to CVS.
- * ext/lame/gstlame.c: (gst_lame_mode_get_type),
- (gst_lame_quality_get_type), (gst_lame_padding_get_type),
- (gst_lame_preset_get_type), (gst_lame_class_init), (gst_lame_init),
- (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup):
- * ext/lame/gstlame.h:
- add preset property to lame so it can use lame presets
+=== release 0.10.11 ===
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/lame/gstlame.c: (gst_lame_get_property):
- whoops forgot break, thanks teuf
+ configure.ac: releasing 0.10.11, "Dumb things"
+ Original commit message from CVS:
+ === release 0.10.11 ===
+ 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ releasing 0.10.11, "Dumb things"
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type),
- (gst_lame_class_init), (gst_lame_src_getcaps),
- (gst_lame_sink_link), (gst_lame_init), (gst_lame_set_property),
- (gst_lame_get_property), (gst_lame_setup):
- * ext/lame/gstlame.h:
- fix lame's broken vbr stuff, allow it to resample if need be, and also
- make xing header optional
+ gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
+ (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
+ Handle the case where an element has multiple pads with
+ unfixed caps as well as still possibly producing more dynamic
+ pads by storing each case as a distinct entry in the dynamic list.
+ Fixes #38223 again.
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_init):
- added getcaps function so samplerate doesnt get fixated to silly values
+ gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ Fix #382223, add more dynamic caps handling.
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
- * ext/lame/gstlame.c: (gst_lame_src_link):
- revert previous fix
+ * po/.gitignore:
+ Ignore all pot files
+ Original commit message from CVS:
+ Ignore all pot files
-2004-08-12 Johan Dahlin <johan@gnome.org>
+2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l/gstv4lelement.c (gst_v4l_iface_supported): Remove bogus
- checks. Doesn't matter what state we are in. Interfaces are a
- compile time thing, not runtime. It also broke the python bindings.
+ gst/audiorate/gstaudiorate.c: Delete bad debug code.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Delete bad debug code.
+ Fixes #381219
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
- * ext/lame/gstlame.c: (gst_lame_src_link):
- made source pad link function check if sinkpad is ok..fixes the problem
- where core fixates the output rate of lame stupidly
+ Fix compilation on win32 under VS8
+ Original commit message from CVS:
+ * gst/videoscale/vs_4tap.c:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ * win32/vs8/libgstvideoscale.vcproj:
+ Fix compilation on win32 under VS8
+ Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
+ Partially fixes #381175
-2004-08-12 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_src_link), (gst_v4lsrc_fixate):
- * sys/v4l/v4l_calls.c:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_set_capture):
- fix fixate function to handle nonsimple caps.
- remove bogus check in _link
- cleanups
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
- * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init):
- set default compression ratio parameter to 0.0 so bitrate parameter
- works :)
+ tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
+ (GST_START_TEST):
+ It would be very bad if, after a discont buffer, we thought every
+ single following buffer was also discont. So, add to the test to
+ ensure that this isn't the case.
+ * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
+ ... it was the case. So fix it.
-2004-08-11 David Schleef <ds@schleef.org>
+2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Improve debug.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (check_queue_event):
+ Improve debug.
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
+ Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
+ padtemplate caps. Refixes #357577.
+
+2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (check_queue_event),
+ (queue_threshold_reached), (queue_out_of_data),
+ (gen_preroll_element):
+ Add event probe to see when EOS is in a queue and we can disable the
+ underrun signals. Fixes #357577.
+
+2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/: New decodebin2 element.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
+ (_gst_boolean_accumulator), (gst_decode_bin_class_init),
+ (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
+ (gst_decode_bin_init), (gst_decode_bin_dispose),
+ (gst_decode_bin_finalize), (gst_decode_bin_set_property),
+ (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
+ (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
+ (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
+ (connect_element), (expose_pad), (type_found),
+ (pad_added_group_cb), (pad_removed_group_cb),
+ (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
+ (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
+ (are_raw_caps), (multi_queue_overrun_cb),
+ (multi_queue_underrun_cb), (gst_decode_group_new),
+ (get_current_group), (group_demuxer_event_probe),
+ (gst_decode_group_control_demuxer_pad),
+ (gst_decode_group_control_source_pad),
+ (gst_decode_group_check_if_blocked),
+ (gst_decode_group_check_if_drained), (gst_decode_group_expose),
+ (gst_decode_group_hide), (gst_decode_group_free),
+ (gst_decode_group_set_complete), (source_pad_blocked_cb),
+ (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
+ (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
+ (plugin_init):
+ New decodebin2 element.
+ Closes #370092
+ * gst/playback/gstplay-marshal.list:
+ Added marshallers for new signals in decodebin2
+ * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
+ Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
+ is set.
+
+2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source),
+ (gst_play_base_bin_change_state):
+ Disable rtsp:// uris for the release, it's not good enough yet.
+ Remove unused var.
+
+2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Implement reverse playback.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (gst_theora_dec_reset),
+ (theora_dec_push_forward), (theora_dec_push_reverse),
+ (theora_handle_data_packet), (theora_dec_decode_buffer),
+ (theora_dec_flush_decode), (theora_dec_chain_reverse),
+ (theora_dec_chain_forward), (theora_dec_chain):
+ Implement reverse playback.
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
+ (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
+ (vorbis_dec_chain_forward):
+ Clear buffers used for reverse playback in _reset.
+ No need to set the eos flag, we clip samples using the segment.
+
+2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Some cleanups.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
+ (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
+ (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
+ (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
+ Some cleanups.
+ Handle continued pages in reverse mode.
+
+2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Small cleanups.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
+ (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
+ (vorbis_dec_flush_decode):
+ Small cleanups.
+ Don't try to add invalid timestamps.
+ Clipping will unref the buffer.
+
+2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/: remove obsolete _factory_init protos
+ Original commit message from CVS:
+ * gst/adder/gstadder.h:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ remove obsolete _factory_init protos
- * gst/tcp/gstfdset.c: Fix compile problem on OS/X.
+2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-08-11 David Schleef <ds@schleef.org>
+ sys/xvimage/xvimagesink.c: Fix spacing in debug message.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
+ Fix spacing in debug message.
- * gst/mpeg1sys/gstmpeg1systemencode.c: Oops, this was correct
- before.
+2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-08-11 David Schleef <ds@schleef.org>
+ ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
+ (gst_ogg_demux_chain):
+ Don't just ignore return values from _pad_push().
+ Small debug improvements.
- * gst-libs/gst/video/videosink.h: Change copyright block to LGPL.
+2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
-2004-08-11 David Schleef <ds@schleef.org>
+ ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
+ If our incoming buffer is marked as DISCONT, then increment the page
+ number (so that the discontinuity is marked in the final ogg
+ bitstream) and flush the previous page.
- * ext/pango/gsttextoverlay.c: Add copyright block and fix plugin
- license field
- * gst-libs/gst/idct/Makefile.am: Remove mmx/sse code
- * gst-libs/gst/video/gstvideosink.c: Change copyright block to
- LGPL.
- * gst/auparse/gstauparse.c: Fix plugin license field.
- * gst/monoscope/gstmonoscope.c: Fix plugin license field.
- * gst/mpeg1sys/gstmpeg1systemencode.c: Fix plugin license field.
- * gst/rtp/gstrtp.c: Fix plugin license field.
+2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
-2004-08-11 Wim Taymans <wim@fluendo.com>
+ ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
+ Original commit message from CVS:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: (gst_theora_enc_init),
+ (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
+ (theora_buffer_from_packet), (theora_enc_is_discontinuous),
+ (theora_enc_chain), (theora_enc_change_state):
+ Mark discontinuities of > 3/4 of a frame, reinit encoder.
+ * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
+ (GST_START_TEST), (theoraenc_suite):
+ Enable discontinuity test, fix it.
- * gst/tcp/Makefile.am:
- * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type), (nearest_pow),
- (ensure_size), (gst_fdset_new), (gst_fdset_free),
- (gst_fdset_set_mode), (gst_fdset_get_mode), (gst_fdset_add_fd),
- (gst_fdset_remove_fd), (gst_fdset_fd_ctl_write),
- (gst_fdset_fd_ctl_read), (gst_fdset_fd_has_closed),
- (gst_fdset_fd_has_error), (gst_fdset_fd_can_read),
- (gst_fdset_fd_can_write), (gst_fdset_wait):
- * gst/tcp/gstfdset.h:
- * gst/tcp/gstmultifdsink.c: (gst_unit_type_get_type),
- (gst_multifdsink_class_init), (gst_multifdsink_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_clear), (gst_multifdsink_get_stats),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property), (gst_multifdsink_init_send),
- (gst_multifdsink_close):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_init), (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_handle_wait), (gst_tcpserversink_init_send),
- (gst_tcpserversink_close):
- * gst/tcp/gsttcpserversink.h:
- Abstracted away the select call, implemented poll (yes we ran into
- the 1024 limit in production).
+2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
+ (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
+ (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
+ (gst_text_overlay_change_state):
+ * ext/pango/gsttextoverlay.h:
+ Some textoverlay fixes: for one, in the video chain function,
+ actually wait for a text buffer to come in if there is none at the
+ moment and there should be one; also, deal more gracefully with
+ incoming buffers that do not have a timestamp or duration; discard
+ text buffer when not needed any longer. Fixes #341681.
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/textoverlay.c:
+ (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
+ (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
+ (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
+ (test_video_waits_for_text_send_text_newsegment_thread),
+ (test_video_waits_for_text_shutdown_element),
+ (test_render_continuity_push_video_buffers_thread),
+ (textoverlay_suite):
+ Add some unit tests for textoverlay.
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcpplugin.c:
- improve debuggging, remove assert
-
-2004-08-10 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_unit_type_get_type),
- (gst_client_status_get_type), (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_remove_client_link),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp-marshal.list:
- Starting to prepare for specifying buffer time in other units
- than buffers. Expose remove reason in signal.
-
-2004-08-10 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_clear),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
- (gst_multifdsink_chain), (gst_multifdsink_close):
- * gst/tcp/gstmultifdsink.h:
- Added more debugging info. Changed the way clients are
- removed from the lists. Fixed a bug where a bad file descriptor
- could cause many clients to be removed.
-
-2004-08-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/videotestsrc/gstvideotestsrc.c: (generate_capslist):
- allow all pixel-aspect-ratios, not just 1:1
-
-2004-08-09 David Schleef <ds@schleef.org>
-
- * sys/glsink/ARB_multitexture.h: Remove old files.
- * sys/glsink/EXT_paletted_texture.h:
- * sys/glsink/NV_register_combiners.h:
- * sys/glsink/gstgl_nvimage.c:
- * sys/glsink/gstgl_pdrimage.c:
- * sys/glsink/gstgl_rgbimage.c:
- * sys/glsink/gstglsink.c:
- * sys/glsink/gstglsink.h:
- * sys/glsink/gstglxwindow.c:
- * sys/glsink/regcomb_yuvrgb.c:
-
-2004-08-09 David Schleef <ds@schleef.org>
-
- Patch from Gernot Ziegler <gz@lysator.liu.se> rewriting the
- GL sink plugin. (Bug #147302)
-
- * configure.ac: Test for OpenGL
- * sys/Makefile.am: Use test for OpenGL
- * sys/glsink/Makefile.am:
- * sys/glsink/glimagesink.c: rewrite
- * sys/glsink/glimagesink.h: rewrite
-
-2004-08-09 David Schleef <ds@schleef.org>
-
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_base_init): Only allow
- sane framerates.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get): same
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support): same
- * testsuite/gst-lint: Test for G_GUINT64_FORMAT usage near gettext.
-
-2004-08-09 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
- Do a bit more logging, make the client_read code more robust.
-
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_init_source),
- (gst_jpegdec_fill_input_buffer), (gst_jpegdec_skip_input_data),
- (gst_jpegdec_resync_to_restart), (gst_jpegdec_term_source),
- (gst_jpegdec_init), (gst_jpegdec_chain):
- * gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
- (gst_multipart_demux_chain), (gst_multipart_demux_change_state):
- cleanups, debugging fixes and memleak plugging
-
-2004-08-09 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (gst_theora_dec_class_init),
- (theora_get_formats), (theora_dec_src_convert),
- (theora_dec_sink_convert), (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_event), (theora_dec_chain),
- (theora_dec_change_state):
- Don't crash on missing header packets.
-
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/LINGUAS:
- * po/sq.po:
- Added Albanian translation (Laurent Dhima)
- * po/cs.po:
- updated
+ gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Avoid integer underflow when the found probability for mp3 is
+ smaller than the 'penalty' we subtract if there's not a clean
+ mp3 header sync at offset 0.
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/lame/gstlame.c:
- fix/add debugging
+ docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add some new symbols to the docs
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- assign all TOO_LAZY's to a real category. Thanks to Warthy Warthog.
+ tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/ffmpegcolorspace.c:
+ (ffmpegcolorspace_suite):
+ Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
+ (for now not for valgrinding though, since it takes too long).
-2004-08-06 Wim Taymans <wim@fluendo.com>
+2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
- Make sure we don't try to read more from a client that what
- ioctl says us or we deadlock.
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps):
+ Fix RGBA32 caps. Fixes #357038.
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_src_link),
- (gst_videotestsrc_change_state), (gst_videotestsrc_src_query),
- (gst_videotestsrc_handle_src_event), (gst_videotestsrc_get):
- decouple running_time and n_frames so it can handle changing
- framerate while running
+ gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Add FIXME so we can add some padding here in 0.11
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/nl.po:
- * po/sv.po:
- updated translations
+ gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ Fix GstBaseRTPAudioPayload structure so the whole GObject
+ inheritance business actually works (parent class instance structure
+ must always come first in the derived class instance structure).
-2004-08-04 Benjamin Otte <otte@gnome.org>
+2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_get_capslist), (generate_capslist),
- (plugin_init):
- generate the list of supported caps at startup and reuse it instead
- of always generating it
-
-2004-07-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/multipart/multipartmux.c: (gst_multipart_mux_pad_link):
- whoops, last checkin broke normal build
-
-2004-08-03 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
- (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
- (gst_alsa_mixer_set_record), (gst_alsa_mixer_set_option),
- (gst_alsa_mixer_get_option):
- * ext/dvdnav/dvdnavsrc.c: (dvdnav_get_event_name),
- (dvdnavsrc_print_event):
- * ext/ogg/gstoggdemux.c: (_find_chain_process), (gst_ogg_print):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_pad_link),
- (gst_ogg_mux_pad_unlink):
- * gst/multipart/multipartmux.c: (gst_multipart_mux_pad_link),
- (gst_multipart_mux_pad_unlink):
- * gst/videofilter/gstvideobalance.c:
- (gst_videobalance_colorbalance_set_value):
- * gst/videomixer/videomixer.c: (gst_videomixer_pad_link),
- (gst_videomixer_pad_unlink):
- * po/uk.po:
- * sys/oss/gstossmixer.c:
- * sys/v4l/gstv4lcolorbalance.c:
- * sys/v4l/gstv4ltuner.c:
- * sys/v4l/v4lsrc_calls.c:
- * sys/v4l2/gstv4l2colorbalance.c:
- * sys/v4l2/gstv4l2tuner.c:
- compile fixes for --disable-gst-debug, G_DISABLE_ASSERT and friends
-
-2004-08-03 Benjamin Otte <otte@gnome.org>
-
- * examples/dynparams/filter.c: (ui_control_create):
- * examples/gstplay/player.c: (print_tag):
- * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
- * ext/gdk_pixbuf/gstgdkanimation.c:
- (gst_gdk_animation_iter_may_advance):
- * ext/jack/gstjack.c: (gst_jack_request_new_pad):
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
- (tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
- * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
- * gst-libs/gst/media-info/media-info-test.c: (print_tag):
- * gst/sine/demo-dparams.c: (main):
- * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- * testsuite/alsa/formats.c: (create_pipeline):
- * testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
- fixes for G_DISABLE_ASSERT and friends
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
- (mp3_type_frame_length_from_header), (mp3_type_find),
- (plugin_init):
- require mp3 typefinding to have at least MIN_HEADERS valid headers
- add typefinding for AAC adts files
-
-2004-08-04 Jan Schmidt <thaytan@mad.scientist.com>
+ Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
+ Original commit message from CVS:
+ * gst/videotestsrc/Makefile.am:
+ * tests/check/Makefile.am:
+ Make sure our checks and the videotestsrc plugin link against the
+ local uninstalled gst libs and not any installed gst libs that
+ might happen to exist as well.
+ * tests/check/elements/adder.c: (message_received),
+ (test_event_message_received), (test_play_twice_message_received):
+ * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
+ Fix compiler warnings when compiling against core with disabled
+ debugging system.
- * sys/ximage/ximagesink.c:
- (gst_ximagesink_calculate_pixel_aspect_ratio):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio):
- Make sure we calculate pixel-aspect-ratio using floating point maths
+2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
+ (gst_audio_rate_sink_event), (gst_audio_rate_chain):
+ Fix audiorate, so that it accurately sets offsets and timestamps.
+ Doesn't change the fundamental algorithmic decisions; so should be
+ safe.
+ * tests/check/Makefile.am:
+ Enable audiorate test now that it passes.
- * po/uk.po:
- updated translation
+2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
+ clear xv when going to NULL, remove // commented non-existant proto
+ * tests/examples/seek/seek.c: (main):
+ add missing tooltip description for scrub and play_scrub
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
- add debugging for display PAR calculation
+2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
-2004-08-02 David Schleef <ds@schleef.org>
+ configure.ac: Bump liboil requirement to 0.3.8.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump liboil requirement to 0.3.8.
+ * gst-libs/gst/riff/riff-media.c:
+ Add Dirac fourcc.
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.h:
+ Use liboil's stdint.h.
+ * gst/videotestsrc/videotestsrc.c:
+ Remove liboil related ifdef's, since they aren't needed now, and
+ won't work with future versions.
- * configure.ac: Fix mikmod CFLAGS.
+2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
+
+ gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
+ Original commit message from CVS:
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ Add a 4-tap image scaler. Theoretically looks much prettier.
+ The tap calculation could use some improvement.
-2004-07-27 Benjamin Otte <otte@gnome.org>
+2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
+
+ Various gsize and gssize printf fixes. Fixes #372507.
+ Original commit message from CVS:
+ Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
+ (gst_riff_parse_strf_iavs):
+ * gst/subparse/gstsubparse.c: (convert_encoding):
+ * gst/tcp/gstmultifdsink.c:
+ (gst_multi_fd_sink_handle_client_write):
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
+ (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
+ (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
+ (gst_ximagesink_ximage_new):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
+ Various gsize and gssize printf fixes. Fixes #372507.
+
+2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
+ (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
+ (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
+ (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
+ (vorbis_dec_chain_forward), (vorbis_dec_chain):
+ * ext/vorbis/vorbisdec.h:
+ First stab at vorbis reverse playback.
+
+2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_event), (gst_base_audio_sink_render):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Make the clock sync code more accurate wrt resampling and playback
+ at different rates.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Use better algorithm to interpolate sample rates.
+
+2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/ogg/gstoggdemux.c: Improve a debug line slightly.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
+ Improve a debug line slightly.
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
+ Call gst_riff_init() in plugin_init, to avoid getting errors from
+ the debug system (unrelated changes to another plugin made this turn
+ up; not sure why).
- * gst/audioscale/gstaudioscale.c:
- - fix templates to only support S16, it's the only format that works
- - make caps nego code use try_set_caps_nonfixed and fixation instead
- of try_set_caps twice, which is not nice for autopluggers
- - change rank to secondary, so autopluggers can pick it up after
- audioconvert
+2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
-2004-08-02 Iain <iain@prettypeople.org>
+ win32/common/libgsttag.def: Add missing symbol (#366492).
+ Original commit message from CVS:
+ Patch by: Sergey Scobich <sergery.scobich at gmail com>
+ * win32/common/libgsttag.def:
+ Add missing symbol (#366492).
- * gst/interleave/interleave.c (interleave_init),
- (interleave_request_new_pad),
- (interleave_pad_removed),
- (interleave_buffered_loop): Use the real pad count, not the artificial
- one.
+2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gststreamselector.c: Don't unref a NULL pad.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
+ Don't unref a NULL pad.
- * configure.ac: bump nano back to development
+2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
-=== release 0.8.3 ===
+ ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
+ (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
+ (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
+ (gst_ogg_demux_loop):
+ Implement first stab at reverse playback.
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac: releasing 0.8.3, "Water"
+ gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ add h263/h264 variants to the caps, Fixes #363118
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio),
- (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_sink_link),
- (gst_xvimagesink_change_state), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- apply similar PAR fixes as to ximagesink
+ gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
+ * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
+ Use g_strerror instead of strerror so we get UTF-8.
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
- patch from: Benjamin Otte
+ ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ Add/remove KW-DIRAC header here, since it is ogg-specific.
- * ext/lame/gstlame.c: (gst_lame_src_link), (gst_lame_init):
- add link function to lame. Fixes #148986.
+2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
-2004-08-02 Johan Dahlin <johan@gnome.org>
+ gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
+ Recognise more mpeg4 elementary video streams.
- * gst/multipart/multipartmux.c (gst_multipart_mux_next_buffer):
- fix debugging log
+2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Lower the probability of mp3 typefinding functions if we don't find a
+ valid mp3 header at the start of the file.
+ Closes #369482
+
+2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (theora_dec_sink_event), (theora_dec_chain_forward),
+ (theora_dec_flush_decode), (theora_dec_chain_reverse),
+ (theora_dec_chain):
+ Document and partially implement an algorithm for doing reverse playback
+ of theora video.
+
+2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
+
+ win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
+ Original commit message from CVS:
+ Patch by: Sergey Scobich <sergey.scobich at gmail com>
+ * win32/common/config.h:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/libgsttag.def:
+ * win32/vs8/gst-plugins-base.sln:
+ * win32/vs8/libgstaudioresample.vcproj:
+ * win32/vs8/libgstinterfaces.vcproj:
+ * win32/vs8/libgstogg.vcproj:
+ * win32/vs8/libgstriff.vcproj:
+ * win32/vs8/libgsttag.vcproj:
+ * win32/vs8/libgsttheora.vcproj:
+ * win32/vs8/libgstvideoscale.vcproj:
+ * win32/vs8/libgstvorbis.vcproj:
+ Misc. VS8 build fixes: fix syntax in config.h, add missing entries
+ to libgsttag.def; add missing dependencies for some vs8 projects;
+ re-arrange placement of .def files in vs8 projects (#366334).
-2004-07-30 David Schleef <ds@schleef.org>
+2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videomixer/Makefile.am: Fix things that should have been
- fixed in the last checkin.
+ ext/ogg/gstogg.c: Remove unused variable.
+ Original commit message from CVS:
+ * ext/ogg/gstogg.c:
+ Remove unused variable.
+ * ext/ogg/gstoggdemux.c:
+ Fix Wim's surname in plugin description.
-2004-07-30 David Schleef <ds@schleef.org>
+2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/multipart/Makefile.am: Fix things that should have been
- fixed in the last checkin.
+ gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
+ Original commit message from CVS:
+ * gst-plugins-base.spec.in:
+ spec new .h file. Fixes #368310.
-2004-07-30 David Schleef <ds@schleef.org>
+2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
- * testsuite/multifilesink/Makefile.am: Fix unused variable.
+ gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
+ (gst_multi_fd_sink_get_stats),
+ (gst_multi_fd_sink_remove_client_link),
+ (gst_multi_fd_sink_queue_buffer),
+ (gst_multi_fd_sink_handle_clients):
+ * gst/tcp/gstmultifdsink.h:
+ Make using the remove or clear signals threadsafe.
+ Make calling get-stats with an invalid fd not segfault.
+ Fixes 368273.
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- bump nano for prerelease
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- updates
+ gst-libs/gst/rtp/: Fix and activate base audio payloader.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_init):
+ Fix and activate base audio payloader.
-2004-07-30 Wim Taymans <wim@fluendo.com>
+2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_clear), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Recover from a select with a bad file descriptor by removing
- the client.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
+ (plugin_init):
+ Add typefinder for QuickTime Image Files (see #366156).
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- fix requirement of core
- * gst-libs/gst/play/play.c: (gst_play_error_plugin),
- (gst_play_pipeline_setup):
- don't use colorspace element. do use hermescolorspace element.
- make macro to get a colorspace element.
- mark strings for translation.
- * po/POTFILES.in:
- add play.c
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- update translations
+ gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
+ Another typo fix (#366212).
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
- fix default for newmedia flag
+ gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_transform_ip):
+ Use stream time to synchronize volume property instead of rather random
+ timestamps. This is needed when gnonlin does its time shifting.
-2004-07-30 Wim Taymans <wim@fluendo.com>
+2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/theora/theoradec.c: (gst_theora_dec_class_init),
- (gst_theora_dec_init), (theora_get_formats),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain), (theora_dec_set_property),
- (theora_dec_get_property):
- * ext/theora/theoraenc.c: (gst_border_mode_get_type),
- (gst_theora_enc_class_init), (gst_theora_enc_init),
- (theora_enc_sink_link), (theora_enc_chain),
- (theora_enc_set_property), (theora_enc_get_property):
- Added cropping option to theora decoder.
- Added border option to theora encoder.
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
- (gst_pngenc_init), (gst_pngenc_chain), (gst_pngenc_get_property),
- (gst_pngenc_set_property):
- * ext/libpng/gstpngenc.h:
- Added newmedia support to pngenc so now gst-launch-0.8 videotestsrc ! ffmpegcolorspace ! pngenc snapshot=false newmedia=true ! multifilesink location=blah%d.png works as expected
+ ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
+ Original commit message from CVS:
+ Patch by: Mark Nauwelaerts <manauw at skynet dot be>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
+ Remove the pad from the element in release_pad.
-2004-07-30 Wim Taymans <wim@fluendo.com>
+2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_enc_chain),
- (theora_enc_set_property), (theora_enc_get_property):
- Fix encoding of non-multiple-of-16 video.
+ sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
+ Explicitly create our custom buffer classes at a thread-safe
+ location as well, since g_type_class_ref() doesn't seem to be
+ entirely thread-safe either (#365501; also see #349410).
-2004-07-29 David Schleef <ds@schleef.org>
+2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: make test for audiofile more strict
+ gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
+ (gst_riff_parse_info):
+ If strings in INFO chunk are not UTF-8, do something similar to
+ what we do for ID3v1 tags: check a number of environment variables
+ (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
+ character sets to try, otherwise try the current locale and/or fall
+ back on ISO-8859-1. Fixes #360552.
-2004-07-25 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_pattern_get_type),
+ (gst_video_test_src_set_pattern):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
+ (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
+ (gst_video_test_src_checkers8):
+ * gst/videotestsrc/videotestsrc.h:
+ Add a bunch of exciting new checkers patterns.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- give different names to typefind functions
+2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
+ Original commit message from CVS:
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect),
+ (gst_sub_parse_format_autodetect), (handle_buffer),
+ (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
+ * gst/subparse/gstsubparse.h:
+ * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
+ (parse_tmplayer):
+ * gst/subparse/tmplayerparse.h:
+ Add support for TMPlayer-type subtitles (#362845).
+ * tests/check/elements/subparse.c: (test_tmplayer_do_test),
+ (GST_START_TEST), (subparse_suite):
+ Add some basic unit tests for the above.
-2004-07-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_calculate_pixel_aspect_ratio),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_getcaps),
- (gst_ximagesink_sink_link), (gst_ximagesink_change_state),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_set_property),
- (gst_ximagesink_get_property), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- allocate PAR's dynamically.
- use autodetected PAR if no object-set PAR is given.
- add workaround for directfb's X not setting physical size.
- fix to xvimagesink will follow tomorrow.
-
-2004-07-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_chain): send tag events downstream
- * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
- (gst_shout2send_get_type), (gst_shout2send_set_clock),
- (gst_shout2send_class_init), (gst_shout2send_init),
- (set_shout_metadata), (gst_shout2send_set_metadata),
- (gst_shout2send_chain), (gst_shout2send_set_property),
- (gst_shout2send_get_property), (gst_shout2send_connect),
- (gst_shout2send_change_state):
- * ext/shout2/gstshout2.h:
- - fix for sending mp3 audio to icecast2 server, if pad link function not
- called before PAUSED state
- - added option to use GStreamer clock sync (as opposed to libshout's own sync)
- - added tagging support for mp3 audio broadcasted
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
- debug info
-
-2004-07-28 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_push):
- Return query failure when we don't know the length of
- an ogg stream insteda of returning TRUE with a bogus value.
-
-2004-07-28 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_get_formats),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- Don't screw up the 1 Chroma for 1 luma sample situation when we
- have an odd offset/width by adding a black border in those cases.
-
-2004-07-28 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_get_formats),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- * ext/theora/theoraenc.c: (theora_enc_sink_link):
- Added first attempt at cropping of the image as required by the
- theora spec. We need more properties in the caps (offset_x,
- offset_y,stride) to implement this correctly.
-
-2004-07-28 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dvdnav/README:
- Update the README to use dvddemux
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
- Ensure getcaps returns a subset of the template caps
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_base_init),
- (gst_mpeg2subt_init):
- Ensure getcaps returns a subset of the template caps
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_class_init),
- (gst_dvd_demux_init), (gst_dvd_demux_get_video_stream),
- (gst_dvd_demux_get_subpicture_stream),
- (gst_dvd_demux_send_subbuffer), (gst_dvd_demux_set_cur_subpicture):
- * gst/mpegstream/gstdvddemux.h:
- Set the explicit caps on the current_video pad before pushing
- anything
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
- (gst_mpeg_demux_get_audio_stream):
- Free caps used to gst_pad_set_explicit_caps, which takes a const
- GstCaps *
-
-2004-07-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: update GStreamer requirement to 0.8.4 because of
- GstFraction.
-
-2004-07-28 Wim Taymans <wim@fluendo.com>
-
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt),
- (gst_wavparse_handle_seek), (gst_wavparse_srcpad_event):
- Add the pad to the element after setting up the caps. This
- makes it a lot easier to autoplug.
-
-2004-07-27 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/median/gstmedian.c:
- * gst/mpeg2subt/gstmpeg2subt.c:
- * gst/mpegaudioparse/gstmpegaudioparse.c:
- * gst/mpegstream/gstdvddemux.c:
- * gst/mpegstream/gstmpegdemux.c:
- * gst/mpegstream/gstmpegpacketize.c:
- * gst/rtjpeg/gstrtjpeg.c:
- * gst/rtjpeg/gstrtjpegdec.c:
- * gst/rtjpeg/gstrtjpegenc.c:
- * gst/sine/gstsinesrc.c:
- * gst/smooth/gstsmooth.c:
- * gst/smpte/gstsmpte.c:
- * gst/smpte/gstsmpte.h:
- * gst/stereo/gststereo.c:
- * gst/videofilter/gstgamma.c:
- * gst/videofilter/gstvideobalance.c:
- * gst/videofilter/gstvideofilter.c:
- * gst/videofilter/gstvideoflip.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/videoscale/videoscale.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/videotestsrc.c:
- * gst/wavenc/gstwavenc.c:
- * gst/wavparse/gstwavparse.c:
- fix local includes and 64 bits constants
+ tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
+ Original commit message from CVS:
+ * tests/check/elements/audiorate.c: (test_injector_base_init),
+ (test_injector_class_init), (test_injector_chain),
+ (test_injector_init), (probe_cb), (do_perfect_stream_test),
+ (GST_START_TEST), (audiorate_suite):
+ More tests for audiorate: inject buffers to check behaviour when
+ buffers overlap.
-2004-07-27 Steve Lhomme <steve.lhomme@free.fr>
+2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
- * win32/gst.sln:
- * gst-libs/gst/*/*.vcproj:
- * gst/*/*.vcproj:
- more working plugins
+ tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
+ (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
+ Add some basic unit tests for audiorate. Disabled at the moment
+ since it doesn't pass yet (see bug #363119).
-2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * testsuite/alsa/Makefile.am:
- * testsuite/alsa/srcstate.c:
- add test for alsasrc changing state
+ gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
+ (parse_subrip), (handle_buffer):
+ Add missing closing tags for markup and fix broken markup,
+ otherwise pango won't render anything (fixes #357531). Also,
+ make sure the text we send out is always NUL-terminated
+ (better safe than sorry etc.).
+ * tests/check/elements/subparse.c: (test_srt_do_test),
+ (test_srt):
+ Some more tests for .srt incl. tests for the above stuff.
-2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
- * gst/silence/gstsilence.c: (gst_silence_init), (gst_silence_link),
- (gst_silence_get):
- * gst/silence/gstsilence.h:
- fix silence generation for 16bit raw audio
+ sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
+ Original commit message from CVS:
+ 2006-10-20 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
+ Patch by: Stefan Kost <ensonic@users.sf.net>
+ Try to redraw borders only when needed. Apparently this consumes
+ resources on small devices... :-O (#363607)
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_metadata),
- (gst_matroska_demux_video_caps), (gst_matroska_demux_plugin_init):
- * gst/mpegaudio/common.c:
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_getcaps), (gst_videoscale_link),
- (gst_videoscale_src_fixate), (gst_videoscale_init),
- (gst_videoscale_finalize):
- * gst/videoscale/gstvideoscale.h:
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_get_capslist):
- * gst/wavenc/gstwavenc.c:
- * sys/oss/gstossmixer.c: (fill_labels):
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_handle_xevents),
- (gst_ximagesink_calculate_pixel_aspect_ratio),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_fixate),
- (gst_ximagesink_getcaps), (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_set_property), (gst_ximagesink_get_property),
- (gst_ximagesink_init), (gst_ximagesink_class_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init):
- * sys/xvimage/xvimagesink.h:
- first batch of pixel aspect ratio commits.
+ gst/tcp/gstmultifdsink.c: If caps change, then update the client's idea of the caps so that we don't end up re-sendin...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c:
+ (gst_multi_fd_sink_client_queue_buffer):
+ If caps change, then update the client's idea of the caps so that we
+ don't end up re-sending streamheaders for every single buffer after
+ the caps change.
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_class_init), (gst_ffmpegcolorspace_chain):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_fill):
- handle stride, needs work if we want to move stride handling
- upstream, but works correctly for our purposes.
-
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videoscale/README:
- add testing examples
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link),
- (gst_videoscale_chain):
- * gst/videoscale/videoscale.c: (gst_videoscale_setup),
- (gst_videoscale_get_size):
- add get_size function that handles stride like videotestsrc.
- fixes conversion for YUV formats for as much as I can test them.
-
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_xvimage_put):
- further cleanups, logging, error handling and synchronizing
-
-2004-07-27 Wim Taymans <wim@fluendo.com>
-
- * gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
- (gst_videomixer_pad_class_init), (gst_videomixer_pad_get_property),
- (gst_videomixer_pad_set_property),
- (gst_videomixer_pad_sinkconnect), (gst_videomixer_pad_init),
- (gst_video_mixer_background_get_type), (gst_videomixer_get_type),
- (gst_videomixer_class_init), (gst_videomixer_init),
- (gst_videomixer_getcaps), (gst_videomixer_request_new_pad),
- (gst_videomixer_blend_ayuv_i420), (pad_zorder_compare),
- (gst_videomixer_sort_pads), (gst_videomixer_fill_checker),
- (gst_videomixer_fill_color), (gst_videomixer_fill_queues),
- (gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
- (gst_videomixer_loop), (plugin_init):
- Be a nicer negotiation citizen and provide a getcaps function on
- the srcpad. This also fixes a crash when resizing.
-
-2004-07-27 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new): Some fixes to image size calculation.
-
-2004-07-27 Wim Taymans <wim@fluendo.com>
-
- * ext/libpng/gstpngdec.c: (gst_pngdec_src_getcaps):
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
- (gst_pngenc_init), (gst_pngenc_chain), (gst_pngenc_get_property),
- (gst_pngenc_set_property):
- * ext/libpng/gstpngenc.h:
- Added snapshot property to pngenc.
- removed g_print from pngdec
-
-2004-07-27 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/ac3parse/ac3parse.vcproj
- * gst/adder/adder.vcproj
- * gst/alpha/alpha.vcproj
- * gst/alpha/alphacolor.vcproj
- * gst/asfdemux/asf.vcproj
- * gst/audioconvert/audioconvert.vcproj
- * gst/audiorate/audiorate.vcproj
- * gst/audioscale/audioscale.vcproj
- * gst/auparse/auparse.vcproj
- * gst/avi/avi.vcproj
- * gst/cdxaparse/cdxaparse.vcproj
- * gst/chart/chart.vcproj
- * gst/colorspace/colorspace.vcproj
- * gst/cutter/cutter.vcproj
- * gst/debug/debug.vcproj
- * gst/debug/efence.vcproj
- * gst/debug/navigationtest.vcproj
- * gst/deinterlace/deinterlace.vcproj
- * gst/effectv/effectv.vcproj
- * gst/ffmpegcolorspace/ffmpegcolorspace.vcproj
- * gst/filter/filter.vcproj
- * gst/flx/flxdec.vcproj
- * gst/goom/goom.vcproj
- * gst/interleave/interleave.vcproj
- * gst/law/alaw.vcproj
- * gst/law/mulaw.vcproj
- * gst/matroska/matroska.vcproj
- * gst/median/median.vcproj
- * gst/mixmatrix/mixmatrix.vcproj
- * gst/mpeg1sys/mpeg1systemencode.vcproj
- * gst/mpeg1videoparse/mp1videoparse.vcproj
- * gst/mpeg2sub/mpeg2subt.vcproj
- * gst/mpegaudio/mpegaudio.vcproj
- * gst/mpegaudioparse/mpegaudioparse.vcproj
- * gst/mpegstream/mpegstream.vcproj
- * gst/multifilesink/multifilesink.vcproj
- * gst/multipart/multipart.vcproj
- * gst/oneton/oneton.vcproj
- * gst/overlay/overlay.vcproj
- * gst/passthrough/passthrough.vcproj
- * gst/qtdemux/qtdemux.vcproj
- * gst/realmedia/rmdemux.vcproj
- * gst/rtjpeg/rtjpeg.vcproj
- * gst/rtp/rtp.vcproj
- * gst/silence/silence.vcproj
- * gst/sine/sinesrc.vcproj
- * gst/smooth/smooth.vcproj
- * gst/smpte/smpte.vcproj
- * gst/spectrum/spectrum.vcproj
- * gst/speed/speed.vcproj
- * gst/stereo/stereo.vcproj
- * gst/switch/switch.vcproj
- * gst/tags/tagedit.vcproj
- * gst/tcp/tcp.vcproj
- * gst/typefind/typefindfunctions.vcproj
- * gst/udp/udp.vcproj
- * gst/videobox/videobox.vcproj
- * gst/videocrop/videocrop.vcproj
- * gst/videodrop/videodrop.vcproj
- * gst/videofilter/gamma.vcproj
- * gst/videofilter/videobalance.vcproj
- * gst/videofilter/videofilter.vcproj
- * gst/videofilter/videoflip.vcproj
- * gst/videoflip/videoflip.vcproj
- * gst/videomixer/videomixer.vcproj
- * gst/videorate/videorate.vcproj
- * gst/videoscale/videoscale.vcproj
- * gst/videotestsrc/videotestsrc.vcproj
- * gst/virtualdub/virtualdub.vcproj
- * gst/volenv/volenv.vcproj
- * gst/volume/volume.vcproj
- * gst/wavenc/wavenc.vcproj
- * gst/wavparse/wavparse.vcproj
- * gst/y4m/y4menc.vcproj
- * gst-libs/gst/audio/audio.vcproj
- * gst-libs/gst/audio/audiofilter.vcproj
- * gst-libs/gst/colorbalance/colorbalance.vcproj
- * gst-libs/gst/idct/idtc.vcproj
- * gst-libs/gst/media-info/media-info.vcproj
- * gst-libs/gst/mixer/mixer.vcproj
- * gst-libs/gst/navigation/navigation.vcproj
- * gst-libs/gst/play/play.vcproj
- * gst-libs/gst/propertyprobe/propertyprobe.vcproj
- * gst-libs/gst/resample/resample.vcproj
- * gst-libs/gst/riff/riff.vcproj
- * gst-libs/gst/tuner/tuner.vcproj
- * gst-libs/gst/video/video.vcproj
- * gst-libs/gst/xoverlay/xoverlay.vcproj
- avoid problems with math.h, fix release dependancy
- rename GStreamer-0.8.lib to libgstreamer.lib
-
-2004-07-27 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate): When
- the atom is not available we have to unlock the mutex. Fixes #148023
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst-libs/gst/media-info/media-info.h:
- issue for a vararg macro with MSVC
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/effectv/effectv.vcproj
- * gst-libs/gst/idct/idct.vcproj:
- * gst-libs/gst/media-info/media-info.vcproj:
- * gst-libs/gst/navigation/navigation.vcproj:
- * gst-libs/gst/propertyprobe/propertyprobe.vcproj:
- * gst-libs/gst/video/video.vcproj:
- * gst-libs/gst/xoverlay/xoverlay.vcproj:
- fixes for build problems
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst-libs/gst/audio/audio.def:
- * gst-libs/gst/audio/riff.def:
- add some definitions needed by plugins
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/asfdemux/gstasfmux.c
- Fix some 64 bits constants to be glib friendly
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/ac3parse/gstac3parse.c
- * gst/audioscale/gstaudioscale.c
- * gst/auparse/gstauparse.c
- * gst/colorspace/gstcolorspace.c
- * gst/colorspace/yuv2rgb.h
- local include fixes
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * win32/gst.sln
- add more plugins to the build
-
-2004-07-26 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new): Some more fixes to image size calculation.
-
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/level/gstlevel.c: (gst_level_link), (gst_level_chain),
- (gst_level_set_property), (gst_level_get_property),
- (gst_level_base_init), (gst_level_class_init):
- add debugging categories. cleanups.
-
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videoscale/videoscale.c: (gst_videoscale_setup),
- (gst_videoscale_planar411), (gst_videoscale_planar400),
- (gst_videoscale_packed422), (gst_videoscale_packed422rev),
- (gst_videoscale_scale_nearest_str1),
- (gst_videoscale_scale_nearest_str2),
- (gst_videoscale_scale_nearest_str4),
- (gst_videoscale_scale_nearest_16bit),
- (gst_videoscale_scale_nearest_24bit):
- fixed stride issues
- tested with 320x240 -> 321, 322, 324 x240
- tested with YV12, I420, YUY2, UYVY
- fixed packed422rev (don't think it could have worked before)
- by testing with UYVY
-
-2004-07-26 Benjamin Otte <otte@gnome.org>
-
- * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_init),
- (gst_lame_chain), (gst_lame_setup), (gst_lame_change_state),
- (plugin_init):
- add debugging category, add error checks like checking return values
- of setup calls, make sure it still works after
- PLAYING=>NULL=>PLAYING, fix encoding of mono streams
-
-2004-07-26 Wim Taymans <wim@fluendo.com>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
- (gst_mpeg_demux_get_audio_stream),
- (gst_mpeg_demux_process_private):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_send_data):
- Check for error codes from the negotiation functions. Make sure
- we really set the pad caps when a new pad is created.
+ ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
+ Original commit message from CVS:
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
+ (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
+ Set caps on pushed buffers; fix up refcounting of caps objects.
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pix_fmt):
- * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_pad_link):
- don't make function do two things at the same time without reason.
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/ac3parse/ac3parse.vcproj
- * gst/adder/adder.vcproj
- * gst/alpha/alpha.vcproj
- * gst/alpha/alphacolor.vcproj
- * gst/asfdemux/asf.vcproj
- * gst/audioconvert/audioconvert.vcproj
- * gst/audiorate/audiorate.vcproj
- * gst/audioscale/audioscale.vcproj
- * gst/auparse/auparse.vcproj
- * gst/avi/avi.vcproj
- * gst/cdxaparse/cdxaparse.vcproj
- * gst/chart/chart.vcproj
- * gst/colorspace/colorspace.vcproj
- * gst/cutter/cutter.vcproj
- * gst/debug/debug.vcproj
- * gst/debug/efence.vcproj
- * gst/debug/navigationtest.vcproj
- * gst/deinterlace/deinterlace.vcproj
- * gst/effectv/effectv.vcproj
- * gst/ffmpegcolorspace/ffmpegcolorspace.vcproj
- * gst/filter/filter.vcproj
- * gst/flx/flxdec.vcproj
- * gst/goom/goom.vcproj
- * gst/interleave/interleave.vcproj
- * gst/law/alaw.vcproj
- * gst/law/mulaw.vcproj
- * gst/matroska/matroska.vcproj
- * gst/median/median.vcproj
- * gst/mixmatrix/mixmatrix.vcproj
- * gst/mpeg1sys/mpeg1systemencode.vcproj
- * gst/mpeg1videoparse/mp1videoparse.vcproj
- * gst/mpeg2sub/mpeg2subt.vcproj
- * gst/mpegaudio/mpegaudio.vcproj
- * gst/mpegaudioparse/mpegaudioparse.vcproj
- * gst/mpegstream/mpegstream.vcproj
- * gst/multifilesink/multifilesink.vcproj
- * gst/multipart/multipart.vcproj
- * gst/oneton/oneton.vcproj
- * gst/overlay/overlay.vcproj
- * gst/passthrough/passthrough.vcproj
- * gst/qtdemux/qtdemux.vcproj
- * gst/realmedia/rmdemux.vcproj
- * gst/rtjpeg/rtjpeg.vcproj
- * gst/rtp/rtp.vcproj
- * gst/silence/silence.vcproj
- * gst/sine/sinesrc.vcproj
- * gst/smooth/smooth.vcproj
- * gst/smpte/smpte.vcproj
- * gst/spectrum/spectrum.vcproj
- * gst/speed/speed.vcproj
- * gst/stereo/stereo.vcproj
- * gst/switch/switch.vcproj
- * gst/tags/tagedit.vcproj
- * gst/tcp/tcp.vcproj
- * gst/typefind/typefindfunctions.vcproj
- * gst/udp/udp.vcproj
- * gst/videobox/videobox.vcproj
- * gst/videocrop/videocrop.vcproj
- * gst/videodrop/videodrop.vcproj
- * gst/videofilter/gamma.vcproj
- * gst/videofilter/videobalance.vcproj
- * gst/videofilter/videofilter.vcproj
- * gst/videofilter/videoflip.vcproj
- * gst/videoflip/videoflip.vcproj
- * gst/videomixer/videomixer.vcproj
- * gst/videorate/videorate.vcproj
- * gst/videoscale/videoscale.vcproj
- * gst/videotestsrc/videotestsrc.vcproj
- * gst/virtualdub/virtualdub.vcproj
- * gst/volenv/volenv.vcproj
- * gst/volume/volume.vcproj
- * gst/wavenc/wavenc.vcproj
- * gst/wavparse/wavparse.vcproj
- * gst/y4m/y4menc.vcproj
- more plugins supported under windows
-
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_ximage_put), (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_alloc):
- Add debugging statements. Use the sizes as returned by the
- *CreateImage calls.
-
-2004-07-26 Johan Dahlin <johan@gnome.org>
-
- * gst/tcp/gsttcpclientsrc.c (gst_tcpclientsrc_get): Make sure that
- the pad is negotiated.
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c (gst_ffmpegcolorspace_chain): Ditto
-
-2004-07-26 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst-libs/gst/colorbalance/colorbalance.vcproj:
- * gst-libs/gst/idct/idct.vcproj:
- * gst-libs/gst/media-info/media-info.vcproj:
- * gst-libs/gst/mixer/mixer.vcproj:
- * gst-libs/gst/navigation/navigation.vcproj:
- * gst-libs/gst/play/play.vcproj:
- * gst-libs/gst/propertyprobe/propertyprobe.vcproj:
- * gst-libs/gst/resample/resample.vcproj:
- * gst-libs/gst/tuner/tuner.vcproj:
- * gst-libs/gst/video/video.vcproj:
- * gst-libs/gst/xoverlay/xoverlay.vcproj:
- more plugins supported under windows
-
-2004-07-25 Iain <iain@prettypeople.org>
-
- * gst/wavparse/gstwavparse.c (gst_wavparse_fmt): Set the caps on the
- pad now rather than when the pad is created because state changes wipe
- explicit caps (fixes #148043).
-
-2004-07-25 Sebastien Cote <sc5@hermes.usherb.ca>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstmad.c:
- fix mad plugin crashing on Sun (fixes #148289)
-
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/avi/avi.def:
- * gst/avi/avi.vcproj:
- * gst/matroska/matroska.def:
- * gst/matroska/matroska.vcproj:
- remove unused .def files
-
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
+ gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
+ (plugin_init):
+ Typefind mmsh header data packet to application/x-mmsh (#362625).
- * gst-libs/gst/audio/gstaudiofilter.c:
- Clean the local include
+2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
+ tests/check/: Add very simple unit test for subparse.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/subparse.c: (buffer_from_static_string),
+ (setup_subparse), (teardown_subparse), (test_srt_do_test),
+ (GST_START_TEST), (subparse_suite):
+ Add very simple unit test for subparse.
- * win32/gst.sln:
- * gst-libs/gst/audio/audio.def:
- * gst-libs/gst/audio/audio.vcproj:
- * gst-libs/gst/audio/audiofilter.vcproj:
- * gst-libs/gst/audio/riff.def:
- * gst-libs/gst/audio/riff.vcproj:
- * gst-libs/gst/gst-libs.def:
- * gst-libs/gst/gst-libs.vcproj:
- * gst/avi/avi.vcproj:
- * gst/avi/avi.vcproj:
- Copy the files where needed after building, cleaner projects
+2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
+ gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
+ (parse_subrip):
+ Strip trailing newlines from subtitle text output.
- * gst/matroska/ebml-write.c:
- Fix some 64 bits constants to be glib friendly
+2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-24 Steve Lhomme <steve.lhomme@free.fr>
+ gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_change_state):
+ Fix memleak; clear subparse->textbuf n state change function.
- * win32/gst.sln:
- * gst-libs/gst/gst-libs.def:
- * gst-libs/gst/gst-libs.vcproj:
- * gst/matroska/matroska.def:
- * gst/matroska/matroska.vcproj:
- Add the preliminary canvas to build plugins on Win32
+2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-23 Benjamin Otte <otte@gnome.org>
+ gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect):
+ Don't require subrip (.srt) files to start with a chunk number of 1.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- don't enfore negotiation from source side, it breaks
- sinesrc ! audioconvert ! osssink
+2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_event), (gst_base_audio_sink_render):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Extract rate from the NEWSEGMENT event.
+ Use commit_full to also take rate adjustment into account when writing
+ samples to the ringbuffer.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Added _commit_full() to also take rate into account.
+ Use simple interpolation algorithm to resample audio.
+ API: gst_ring_buffer_commit_full()
+ * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
+ * tests/examples/seek/seek.c: (segment_done):
+ Don't try to seek with 0.0 rate, just pause instead.
+ Remove bogus debug line.
+
+2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
+ (setup_source):
+ Catch async errors when starting up the subtitle bin, so we can
+ stop waiting and continue with the main film instead of hanging
+ forever. Fixes #339366.
+ * tests/check/elements/playbin.c: (playbin_suite):
+ Enable unit test for the above.
-2004-07-22 David Schleef <ds@schleef.org>
+2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/playbin.c: (GST_START_TEST),
+ (gst_red_video_src_uri_get_type),
+ (gst_red_video_src_uri_get_protocols),
+ (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
+ (gst_red_video_src_uri_handler_init),
+ (gst_red_video_src_init_type), (gst_red_video_src_base_init),
+ (gst_red_video_src_create), (gst_red_video_src_class_init),
+ (gst_red_video_src_init), (plugin_init), (playbin_suite):
+ Some small and basic unit tests for playbin; not very useful yet,
+ but at least a start.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add typefind
- for ELF files, since they can easily be recognized as audio/mpeg.
- (bug #147441)
+2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gstplaybin.c: The old pad activation spiel.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (setup_sinks):
+ The old pad activation spiel.
- * gst/videoscale/videoscale.c: (gst_videoscale_setup),
- (gst_videoscale_planar411), (gst_videoscale_scale_nearest_32bit),
- (gst_videoscale_scale_nearest_24bit),
- (gst_videoscale_scale_nearest_16bit):
- fix 16bit and 24bit for stride (24bit might need testing)
- don't pretend we do more than one algorithm
+2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Don't hang forever if the subbin already fails to start up in
+ the state change to PAUSED (#339366).
- * configure.ac:
- * gst/Makefile.am:
- * gst/multifilesink/Makefile.am:
- * gst/multifilesink/gstmultifilesink.c:
- (gst_multifilesink_get_formats),
- (gst_multifilesink_get_query_types), (_do_init),
- (gst_multifilesink_base_init), (gst_multifilesink_class_init),
- (gst_multifilesink_init), (gst_multifilesink_dispose),
- (gst_multifilesink_set_location), (gst_multifilesink_set_property),
- (gst_multifilesink_get_property), (gst_multifilesink_open_file),
- (gst_multifilesink_close_file), (gst_multifilesink_next_file),
- (gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
- (gst_multifilesink_chain), (gst_multifilesink_change_state),
- (gst_multifilesink_uri_get_type),
- (gst_multifilesink_uri_get_protocols),
- (gst_multifilesink_uri_get_uri), (gst_multifilesink_uri_set_uri),
- (gst_multifilesink_uri_handler_init), (plugin_init):
- * gst/multifilesink/gstmultifilesink.h:
- * testsuite/Makefile.am:
- * testsuite/multifilesink/Makefile.am:
- * testsuite/multifilesink/fakesrc_test.c: (gst_newmedia_base_init),
- (gst_newmedia_class_init), (gst_newmedia_init),
- (gst_newmedia_chain), (gst_newmedia_trigger), (test_format),
- (newfile_signal), (test_signal), (main):
- multifilesink plugin for creating new files every time a new media
- discontinuity event occurs
-
-2004-07-22 Wim Taymans <wim@fluendo.com>
-
- * gst/alpha/Makefile.am:
- * gst/alpha/gstalphacolor.c: (gst_alpha_color_get_type),
- (gst_alpha_color_base_init), (gst_alpha_color_class_init),
- (gst_alpha_color_init), (gst_alpha_color_set_property),
- (gst_alpha_color_get_property), (gst_alpha_color_sink_link),
- (transform), (gst_alpha_color_chain),
- (gst_alpha_color_change_state), (plugin_init):
- Stupid plugin to to RGBA to AYUV conversion because none of
- the colorspace plugins can handle that yet.
-
-2004-07-22 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (update_scale), (main):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_is_dynamic),
- (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (no_more_pads), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (queue_overrun),
- (gen_preroll_element), (remove_prerolls), (unknown_type),
- (no_more_pads), (new_stream), (setup_source),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (play_base_eos), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- * gst/playback/gststreaminfo.h:
- More playback updates, attempt to fix things after the state change
- breakage.
+2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
+ (gst_tuner_set_channel), (gst_tuner_get_channel),
+ (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
+ (gst_tuner_set_frequency), (gst_tuner_get_frequency),
+ (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
+ (gst_tuner_find_channel_by_name):
+ Fix some function guards, add some more function guards.
- * gst/videoscale/videoscale.c: (gst_videoscale_planar411),
- (gst_videoscale_scale_nearest_16bit):
- comment algorithm
+2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
+ (remove_element_chain):
+ Don't return a pad from get_our_ghost_pad unless it is actually the
+ one we want.
+ Change a cast in remove_element_chain slightly.
+
+2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
+ Original commit message from CVS:
+ 2006-10-13 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek), (start_seek),
+ (rate_spinbutton_changed_cb), (segment_done),
+ (msg_state_changed):
+ Segment seeking needs to use the rate and set stop to -1.
+
+2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_setcaps):
+ Don't crash when ringbuffer is not yet created.
+ Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
+ Fixes #361634.
+ * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_request_new_pad):
+ Activate pads befre adding them to running elements.
+
+2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
+ Original commit message from CVS:
+ 2006-10-13 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek), (start_seek),
+ (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
+ scale
+ updater when we start grabing the slider. Don't wait for the
+ pipeline to be PAUSED.
+
+2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
+ (gst_mixer_set_volume), (gst_mixer_get_volume),
+ (gst_mixer_set_mute), (gst_mixer_set_option),
+ (gst_mixer_get_option), (gst_mixer_mute_toggled),
+ (gst_mixer_record_toggled), (gst_mixer_volume_changed),
+ (gst_mixer_option_changed):
+ Guard mixer interface functions against bogus arguments.
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
+ Original commit message from CVS:
+ 2006-10-12 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek), (start_seek),
+ (stop_seek),
+ (play_cb), (pause_cb), (stop_cb),
+ (rate_spinbutton_changed_cb),
+ (msg_state_changed), (main): Use state-changed messages to
+ trigger
+ start/stop of scale update timer. Indeed the scale slider was
+ jumping here and there because the update timer was activated
+ before seek completed. This fixes instant applying of rate
+ changes
+ by pressing the spinbutton like a crazy man !
+
+2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
+ Original commit message from CVS:
+ Patch by: Sebastien Cote <sebas642 at yahoo.ca>
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
+ (gst_basertppayload_finalize):
+ Fix two small memory leaks (#361456).
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_src_link),
- (gst_videotestsrc_init), (gst_videotestsrc_get),
- (gst_videotestsrc_set_pattern), (gst_videotestsrc_set_property),
- (gst_videotestsrc_get_property):
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- cleanup and commenting
-
-2004-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
- (gst_ogg_demux_get_formats), (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_demux_src_convert),
- (gst_ogg_demux_handle_event), (gst_ogg_demux_seek_before),
- (_find_chain_get_unknown_part), (_find_streams_check),
- (gst_ogg_demux_push), (gst_ogg_pad_push):
- * ext/theora/theoradec.c: (theora_get_formats),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_event):
- More seeking fixes, oggdemux now supports seeking to time and
- uses the downstream element to convert granulepos to time.
- Seeking in theora-only ogg files now works.
-
-2004-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (theora_get_formats), (theora_get_event_masks),
- (theora_get_query_types), (theora_dec_src_convert),
- (theora_dec_sink_convert), (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_event), (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (vorbis_get_event_masks), (vorbis_get_query_types),
- (gst_vorbis_dec_init), (vorbis_dec_convert),
- (vorbis_dec_src_query), (vorbis_dec_src_event), (vorbis_dec_event):
- Added query/convert/formats functions to vorbis and theora decoders
- so that the outside world can use them too. Fixed seeking on an
- ogg/theora/vorbis file by disabling the seeking seeking on the
- theora srcpad.
-
-2004-07-21 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_set_xwindow_id): Optimize
- images creation for both elements. We don't create the image on caps
- nego or renego, we just destroy the internal one if present if it does
- not match the needs. The chain function takes care of creating a new
- image when needed.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xwindow_decorate), (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_xwindow_id): Additionally xvimage now contains
- the image format information. The buffer pool checks for the context
- image format and discard images with different formats.
- * sys/xvimage/xvimagesink.h: Adding im_format in the xvimage structure.
-
-2004-07-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
+ Original commit message from CVS:
+ 2006-10-10 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek),
+ (rate_spinbutton_changed_cb): When changing spinbutton we try
+ to change the rate on the fly.
+
+2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/: Add WMS caps.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Add WMS caps.
+
+2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
+
+ ext/gnomevfs/: Fix URI interface implementation return type.
+ Original commit message from CVS:
+ 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Patch by: Josep Torre Valles <josep@fluendo.com>
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Fix URI interface implementation return type.
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
+ Fix what looks like a copy/paste issue when assigning values.
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_get_type):
+ Cast to prevent Forte warnings.
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ Fix URI interface implementation return type.
+ gst_pad_query_position requires a signed integer pointer as
+ 3rd parameter, GstClockTime is unsigned.
+ * gst/audioconvert/audioconvert.c:
+ Fix integer overflow when treated as signed.
+ * gst/audioresample/resample.c: (resample_add_input_data):
+ Cast to prevent warnings on Forte.
+ * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
+ Fix integer overflow when treated as signed.
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
+ * gst/playback/gstdecodebin.c: (queue_filled_cb),
+ (cleanup_decodebin):
+ Who initialises a guint to -1!
+ Cast function pointers to prevent warnings on Forte.
+ * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
+ (queue_threshold_reached):
+ Cast function pointers correctly to prevent warnings on Forte.
+ * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
+ Cast function pointers correctly to prevent warnings on Forte.
+ * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
+ Obvious change to unsigned, 0xEF > max signed char.
+ * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
+ GstClockTime is unsigned, initialise correctly.
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
+ Cast so pointer arithemetic doesn't cause warnings on Forte.
+ * gst/videorate/gstvideorate.c:
+ Use correct return value.
+ * tests/examples/seek/scrubby.c:
+ GstClockTime is unsigned, initialise correctly.
+
+2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
+
+ gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
+ Original commit message from CVS:
+ Patch by: Ferenc Gerlits <fgerlits at gmail com>
+ * gst/typefind/gsttypefindfunctions.c:
+ Recognise XML files and XML-like files shorter than 256 bytes as
+ well (fixes #359237).
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- no point in doing any chaining if the pad we want to push from
- isn't usable.
+2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
-2004-07-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * common:
+ * gst/typefind/gsttypefindfunctions.c:
+ Added typefind functions to video/x-nuv media.
+ Original commit message from CVS:
+ Added typefind functions to video/x-nuv media.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Fix double end-to-native symbol conversion (#148021).
+2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-20 David Schleef <ds@schleef.org>
+ gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
+ Some more guards against invalid input.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate):
- Don't use an Atom that doesn't exist.
+2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ ext/pango/gsttextoverlay.c: Useless goto.
+ Original commit message from CVS:
+ 2006-10-07 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
+ Useless goto.
+ * tests/examples/seek/seek.c: (do_seek),
+ (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
+ seek example to experiment with rates != 1.0 (reverse playback
+ !)
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer):
- * gst/tcp/gstmultifdsink.h:
- More multifdsink stats. Avoid deadlock by releasing locks
- before sending out a signal.
+2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ Unref message in doc-example (spotted by Robert McQueen)
- * po/LINGUAS:
- * po/hu.po:
- added Hungarian translation (Laszlo Dvornik)
+2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ gst/typefind/gsttypefindfunctions.c: printf fix.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (mpeg1_parse_header), (mpeg1_sys_type_find):
+ printf fix.
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer):
- * gst/tcp/gsttcp-marshal.list:
- Fixed the stupid marshal definition.
+2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ gst/playback/: Activate dynamic pads before adding them to the element.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
+ (close_pad_link):
+ * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
+ Activate dynamic pads before adding them to the element.
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_chain),
- (gst_multifdsink_set_property), (gst_multifdsink_get_property),
- (gst_multifdsink_init_send):
- * gst/tcp/gstmultifdsink.h:
- Added more stats, added timeout for a client, fixed some typos
- and added some comments.
+2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
+ Original commit message from CVS:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Fix obviously-bogus macros; use the correct types.
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp-marshal.list:
- Added get_stats method that returns a GValueArray of
- stats values.
+2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-19 Benjamin Otte <otte@gnome.org>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_change_state):
+ Also call parent state change function to activate pads.
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (mpeg1_parse_header), (mpeg1_sys_type_find):
+ Add some more debug info in mpeg typefinding.
- * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
- make sure longname, description and author are valid UTF-8
+2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
-2004-07-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_chain):
+ Zero byte theora packets are valid and well-defined; don't warn on
+ them.
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state),
- (gst_ximagesink_set_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_property):
- make sure SYNCHRONOUS is respected after getting the X context
+2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_get_stats), (find_limits),
+ (gst_multi_fd_sink_queue_buffer):
+ API: add dropped_buffers to the get-stats GValueArray
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_src_event),
- (gst_matroska_demux_parse_blockgroup):
- * gst/matroska/matroska-ids.h:
- add BlockReference tag and ignore it to clear out log.
- ignore NAVIGATION events to clear out log.
+2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ Printf format fixes.
+ Original commit message from CVS:
+ * ext/alsa/gstalsadeviceprobe.c:
+ (gst_alsa_device_property_probe_get_values):
+ * ext/alsa/gstalsasink.c: (set_hwparams):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
+ (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
+ (gst_ogg_mux_process_best_pad):
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
+ (gst_ogg_parse_chain):
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
+ (gst_vorbis_enc_buffer_check_discontinuous):
+ * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_handle_track_seek):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_push_full):
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ * gst/audioresample/resample.c: (resample_input_pushthrough):
+ * gst/playback/gstplaybasebin.c: (queue_out_of_data):
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (wavpack_type_find):
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ Printf format fixes.
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
- (gst_matroska_demux_add_stream):
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init):
- add debug categories
+2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-16 Wim Taymans <wim@fluendo.com>
+ gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
+ Original commit message from CVS:
+ * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
+ Fix a simple mistake (see the docs)
+ Fixes #359580
- * ext/libpng/Makefile.am:
- * ext/libpng/gstpng.c: (plugin_init):
- * ext/libpng/gstpngdec.c: (user_error_fn), (user_warning_fn),
- (gst_pngdec_get_type), (gst_pngdec_base_init),
- (gst_pngdec_class_init), (gst_pngdec_sinklink), (gst_pngdec_init),
- (gst_pngdec_src_getcaps), (user_read_data), (gst_pngdec_chain):
- * ext/libpng/gstpngdec.h:
- Added png decoder.
+2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-16 Julien MOUTTE <julien@moutte.net>
+ * win32/common/config.h:
+ bump version
+ Original commit message from CVS:
+ bump version
+
+2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Add vorbistag element to docs; update version numbers to 0.10.10.1.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xerror),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new),
- (gst_ximagesink_ximage_destroy), (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_free),
- (gst_ximagesink_buffer_alloc):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.h: Getting the 2 video sinks synchronized
- again. Using internal data pointer of the x(v)image to store image's
- data to be coherent with the buffer alloc mechanism. Investigated the
- image destruction code to be sure that everything gets freed correctly.
+2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
+
+ ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
+ Original commit message from CVS:
+ Patch by: James "Doc" Livingston <doclivingston at gmail com>
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/vorbis.c: (plugin_init):
+ * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
+ (vorbis_parse_parse_packet), (vorbis_parse_chain):
+ * ext/vorbis/vorbisparse.h:
+ * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
+ (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
+ (gst_vorbis_tag_parse_packet):
+ * ext/vorbis/vorbistag.h:
+ Add new vorbistag element which derives from vorbisparse
+ and is essentially the same as well, only that it implements
+ the GstTagSetter interface and can modify the stream's
+ vorbiscomment on the fly (#335635).
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/vorbistag.c: (setup_vorbistag),
+ (cleanup_vorbistag), (buffer_probe), (start_pipeline),
+ (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
+ (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
+ Add unit test for new vorbistag element.
-2004-07-16 Wim Taymans <wim@fluendo.com>
+2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data),
- (gst_riff_read_strf_auds_with_data):
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- Make sure we don't create 0 sized subbuffers in riff-read.
- Signal the no more pads signal after reading the avi header.
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_is_dynamic),
- (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (no_more_pads), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (gst_decode_bin_change_state), (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (queue_overrun),
- (gen_preroll_element), (remove_prerolls), (unknown_type),
- (no_more_pads), (new_stream), (setup_source),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (play_base_eos), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- Better error recovery. Added configurable preroll queue size. Faster
- detection of no-more-pads.
+ ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
+ (vorbis_parse_push_headers), (vorbis_parse_chain):
+ Set BOS flag in packet structure to fix 'jump depends
+ on unitialized value' errors in valgrind; various minor
+ clean-ups.
-2004-07-16 Wim Taymans <wim@fluendo.com>
+2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstdecodebin.c: Fix typo in a debug statement.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ Fix typo in a debug statement.
+ * gst/playback/gstplaybasebin.c: (probe_triggered),
+ (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
+ (gen_source_element), (source_new_pad), (analyse_source),
+ (setup_source):
+ When handling no_more_pads in new_decoded_pad, make sure to treat
+ subtitle pads correctly. Fixes playback with subtitle files.
+ Move a recurring message to LOG level.
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
+ which ends up as -1 when cast to an int. Make the logic handle the
+ max value as an unsigned mask and only change the colorkey when it's
+ a value we recognise.
+
+2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ Removed empty * between paragraphs
+
+2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/: Moved some documentation into .c file
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/README:
+ Moved some documentation into .c file
+
+2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Fix compilation.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (no_more_pads):
+ Fix compilation.
+
+2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Remove g_print
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (new_caps):
+ Remove g_print
+ * gst/playback/gstplaybin.c:
+ Add some docs.
+
+2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Re-enable cddabasesrc test to see if it works again
+ now.
- * gst-libs/gst/video/video.h:
- Added 32 bits RGBA. Not sure if we should use another mime-type
- for alpha rgb. Currently the presence of the alpha_mask property
- signals an alpha channel.
+2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-16 Wim Taymans <wim@fluendo.com>
+ gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_subtitle),
+ (gen_source_element):
+ Handle invalid URIs a bit more gracefully.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- FPS seems to be 0.0 to MAX everywhere else.
+2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/pipelines/oggmux.c: Remove obsolete comment.
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c:
+ Remove obsolete comment.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- mp42/mp43 (no caps) exist too.
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
- Set pixel_width/height; we've got them in-caps.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- * gst/wavparse/gstwavparse.c: (plugin_init):
- Both are valid primary.
- * sys/oss/gstossmixer.c:
- Remove i18n hack and enable translations.
+2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
-2004-07-15 Benjamin Otte <otte@gnome.org>
+ ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
+ (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
+ (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
+ (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
+ (gst_ogg_mux_collected):
+ Commit patch from James "Doc" Livingston, adds proper EOS handling
+ in oggmux. GStreamer can, for the first time ever, create a valid
+ Ogg file! Yay!
+ * tests/check/pipelines/oggmux.c: (check_chain_final_state),
+ (oggmux_suite):
+ Reenable tests now that they pass.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy):
- fix for non-shm xv. Original patch by Tim Ringenbach (fixes #147248)
+2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
+ Stop reading commands when EOF (we read 0) as well.
+
+2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
+ (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
+ (find_dynamic), (unlinked), (close_link):
+ Implement delayed caps linking needed for element with a lot of
+ different caps on the src pads that get fixed at runtime.
+ Improve management of dynamic elements.
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
+ (group_destroy), (group_commit), (check_queue), (queue_overrun),
+ (gen_preroll_element), (remove_groups), (unknown_type),
+ (add_element_stream), (no_more_pads_full), (no_more_pads),
+ (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
+ (new_decoded_pad), (setup_subtitle), (array_has_value),
+ (gen_source_element), (source_new_pad), (has_all_raw_caps),
+ (analyse_source), (remove_decoders), (make_decoder),
+ (remove_source), (setup_source), (finish_source), (prepare_output),
+ (gst_play_base_bin_change_state):
+ * gst/playback/gstplaybasebin.h:
+ Use more _CAST instead of full type checking casts.
+ Small cleanups, plug some leaks.
+ Handle dynamic sources.
+ Add some helper functions to create lists of strings used for
+ blacklisting and other stuff.
+ Refactor some code dealing with analysing the source.
+ Re-enable sources without pads (like cd:// or other selfcontained
+ elements).
+
+2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ When we have a timestamp, we can still perform clipping.
+ When we have no clock, we must play the sample ASAP.
+
+2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Set caps on outgoing buffers.
+ * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
+ (gst_video_rate_event), (gst_video_rate_chain):
+ * gst/videorate/gstvideorate.h:
+ Fix videorate some more. Fixes #357977
+
+2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
+ Original commit message from CVS:
+ * tests/check/elements/adder.c: (adder_suite):
+ Don't set timeout to 6 seconds when we're running
+ in valgrind ... (and how is 6 seconds longer than
+ the default anyway?)
-2004-07-15 Benjamin Otte <otte@gnome.org>
+2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio),
- (gst_alsa_sw_params_dump), (gst_alsa_hw_params_dump),
- (gst_alsa_close_audio):
- disable some of the debugging code for now. Writing debugging to a
- buffer is broken in current alsalib releases.
+ gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
+ (gst_audio_rate_sink_event), (gst_audio_rate_convert),
+ (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
+ Keep sink and src segment to keep track of time and support more
+ input formats.
+ Fix bogus next_offset and run_time calculation, don't understand how
+ this could have worked before. Fixes #357976.
+ Remove some unneeded vars.
-2004-07-12 Benjamin Otte <otte@gnome.org>
+2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_alloc_buffer):
- use bufferpools
+ gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (remove_sinks):
+ Only remove visualisation from visbin if there is a visbin (or:
+ don't throw warnings when closing totem without playing a file).
-2004-07-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Add some more info in a WARNING.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Handle PAUSE in create function, use new -core addition to
+ wait for playing. Fixes pausing and resuming capture from an
+ audiosrc.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ Constify some more.
+ Caller supports interrupted reads now.
+
+2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
- * ext/theora/theoradec.c: (gst_theora_dec_class_init),
- (theora_dec_src_query), (theora_dec_event):
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
- add debugging categories. Remove \n's.
+ * gst-plugins-base.spec.in:
+ add new header file to spec
+ Original commit message from CVS:
+ add new header file to spec
-2004-07-13 Johan Dahlin <johan@gnome.org>
+2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybin.c (gst_play_bin_set_property)
- (gst_play_bin_get_property): Impl.
+ tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Another attempt to make the gen64 buildbot happy.
-2004-07-13 Wim Taymans <wim@fluendo.com>
+2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_seek_before):
- When trying to find the stream length, seek back N pages
- instead of just one, where N is the number of streams in
- the current chain.
+ ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
+ Original commit message from CVS:
+ Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
+ * ext/libvisual/visual.c: (gst_visual_clear_actors),
+ (gst_visual_chain), (gst_visual_change_state):
+ Libvisual plugin was not passing audio data to libvisual 0.4.0
+ correctly. Fixes #357800
-2004-07-13 Wim Taymans <wim@fluendo.com>
+2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data),
- (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data),
- (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_auds):
- * gst-libs/gst/riff/riff-read.h:
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_add_stream):
- Set codec_data on caps for avidemuxer.
-
-2004-07-12 David Schleef <ds@schleef.org>
-
- * configure.ac: Fix test for Objective C
-
-2004-07-12 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_get_capslist),
- (gst_gdk_pixbuf_chain):
- Add svg and pcx to template caps, and ensure that getcaps returns a
- subset of the template caps.
- Copy each row manually for output, as gdkpixbuf may pad the
- rowstride to a 32-bit word boundary.
-
-2004-07-12 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_video_template_caps):
- Fix the template caps to include some more media types.
-
-2004-07-12 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (print_feature), (gst_decode_bin_init),
- (gst_decode_bin_dispose), (find_compatibles), (close_pad_link),
- (try_to_link_1), (new_pad), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (gst_decode_bin_change_state), (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (queue_overrun),
- (gen_preroll_element), (remove_prerolls), (no_more_pads),
- (new_stream), (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_get_property), (play_base_eos),
- (gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gen_video_element),
- (gen_audio_element), (remove_sinks), (setup_sinks),
- (gst_play_bin_change_state), (gst_play_bin_get_event_masks),
- (gst_play_bin_send_event), (gst_play_bin_get_formats),
- (gst_play_bin_convert), (gst_play_bin_get_query_types),
- (gst_play_bin_query), (plugin_init):
- * gst/playback/test4.c: (main):
- More fixes on reusing of the element.
-
-2004-07-11 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstmad.c: (normal_seek):
- allow seeking for other methods than just SET
-
-2004-07-11 Andy Wingo <wingo@pobox.com>
-
- * gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For
- float, "any" caps -> buffer_frames=[0,MAX].
-
- * gst/interleave/interleave.c (interleave_getcaps): Seems the core
- doesn't intersect our caps with the template any more. Do it
- ourselves.
- (interleave_buffered_loop): Use g_newa instead of malloc/free.
-
-2004-07-09 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (print_feature), (gst_decode_bin_init),
- (gst_decode_bin_dispose), (find_compatibles), (close_pad_link),
- (try_to_link_1), (new_pad), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (gst_decode_bin_change_state), (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (queue_overrun),
- (gen_preroll_element), (remove_prerolls), (no_more_pads),
- (new_stream), (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_get_property), (play_base_eos),
- (gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gen_video_element),
- (gen_audio_element), (remove_sinks), (setup_sinks),
- (gst_play_bin_change_state), (gst_play_bin_get_event_masks),
- (gst_play_bin_send_event), (gst_play_bin_get_formats),
- (gst_play_bin_convert), (gst_play_bin_get_query_types),
- (gst_play_bin_query), (plugin_init):
- * gst/playback/test4.c: (main):
- Work on object reuse and seeking.
-
-2004-07-09 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (iterate):
- Don't consume all CPU in the idle loop.
-
-2004-07-09 Wim Taymans <wim@fluendo.com>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_new_output_pad),
- (gst_mpeg_demux_parse_packet), (gst_mpeg_demux_process_private):
- Add pad to element *after* setting the pad functions so that
- the scheduler can use the correct ones.
-
-2004-07-09 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_dec_from_granulepos),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_chain):
- Sync to keyframe after seek
-
-2004-07-09 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
- (gst_alsa_sink_loop), (gst_alsa_sink_change_state):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_change_state):
- * ext/libvisual/visual.c: (gst_visual_change_state):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_change_state):
- * ext/theora/theoradec.c: (theora_dec_change_state):
- * ext/theora/theoraenc.c: (theora_enc_change_state):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
- * gst-libs/gst/navigation/navigation.c:
- * gst/adder/gstadder.c: (gst_adder_change_state):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_get_buffer):
- * gst/multipart/multipartdemux.c:
- (gst_multipart_demux_change_state):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- * gst/volume/gstvolume.c: (volume_chain_int16):
- don't assert in state change, this should be done by the base
- GstElement class.
- various debugging fixes.
+ tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
+ Original commit message from CVS:
+ * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
+ Add timeout to _get_state() so we see which pipeline it is
+ that causes trouble on the gen64 build bot.
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
+ (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ the source pad always uses fixed caps.
+
+2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Added docs for the audio libs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Added docs for the audio libs.
+
+2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Temporarily disable test that fails on the bots for unknown reasons.
- * configure.ac:
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_dispose), (gst_play_set_location),
- (gst_play_set_data_src), (gst_play_set_video_sink),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_sink_element),
- (gst_play_get_all_by_interface):
- * gst-libs/gst/play/play.h:
- add new method to get elements implementing an interface.
- add various error logging
-
-2004-07-08 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_dv_pipeline), (make_avi_pipeline),
- (make_mpeg_pipeline), (make_mpegnt_pipeline),
- (make_playerbin_pipeline), (query_durations_elems),
- (query_durations_pads), (query_positions_elems),
- (query_positions_pads), (update_scale), (iterate), (stop_seek),
- (main):
- Added playbin seeking example.
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * gst-libs/gst/play/play.c: (gst_play_set_location),
- (gst_play_set_data_src), (gst_play_set_video_sink),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_framerate):
- use a macro to look up elements from hash table
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_get_length_callback), (gst_play_set_location),
- (gst_play_seek_to_time), (gst_play_set_data_src),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization), (gst_play_connect_visualization),
- (gst_play_get_sink_element):
- - add debugging info
- - fix looking up sink elements by iterating over complete caps
- - put everything except for source and autoplugger in a complete bin
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_drain_audio):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_flush_one_pad),
- (gst_alsa_sink_check_event), (gst_alsa_sink_mmap),
- (gst_alsa_sink_write), (gst_alsa_sink_loop):
- * ext/alsa/gstalsasink.h:
- - add debugging info
- - clean up schizophrenia of data/buffer/event
- - fix double event unref error
+2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ Moved AudioCodecType into priv
+ Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
+
+2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
+ (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
+ (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
+ (new_pad):
+ Cleanups and small leak fixes.
+ Added Depayloaders to valid list of autopluggable elements.
+
+2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
+ (gen_video_element), (gen_text_element), (gen_audio_element),
+ (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
+ (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
+ Detect NO_PREROLL state change returns and disable clock distribution to
+ the sinks so that sync is disabled.
+ Avoid some type checking and do simple casts instead.
+ Small cleanups, fix some FIXMEs.
+ Be more robust when linking user specified elements, catch an report
+ errors. Fixes #357404.
+ Fix some leaks in the error paths.
+
+2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-08 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ ChangeLog surgery for missing bug-number
+ Original commit message from CVS:
+ ChangeLog surgery for missing bug-number
- * gst/playback/Makefile.am:
- Add headers to noinst
+2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/playback/test.c:
+ Fix compilation with uClibc and -Werror (#357591).
- * tools/gst-launch-ext-m.m:
- * tools/gst-launch-ext.1.in:
- convert to the third millenium
+2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 David Schleef <ds@schleef.org>
+ gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ Parse dates that are followed by a time as well (#357532).
+ * tests/check/libs/tag.c: (test_vorbis_tags):
+ Add unit test for this.
- * sys/dxr3/Makefile.am: noinst_SOURCES should be nodist_SOURCES
+2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ gst/: A few array const-ifications.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
+ (gst_audio_convert_transform_caps):
+ * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
+ * gst/videotestsrc/videotestsrc.h:
+ A few array const-ifications.
- * gst/playback/Makefile.am:
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (print_feature), (gst_decode_bin_init),
- (gst_decode_bin_dispose), (find_compatibles), (close_pad_link),
- (try_to_link_1), (new_pad), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (rebuild_pipeline), (queue_overrun),
- (gen_preroll_element), (no_more_pads), (new_stream),
- (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_get_property), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gen_video_element),
- (gen_audio_element), (setup_sinks), (gst_play_bin_change_state),
- (gst_play_bin_get_event_masks), (gst_play_bin_send_event),
- (gst_play_bin_get_formats), (gst_play_bin_convert),
- (gst_play_bin_get_query_types), (gst_play_bin_query),
- (plugin_init):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (gst_stream_info_get_type), (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (gst_stream_info_set_property),
- (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (main):
- * gst/playback/test2.c: (main):
- * gst/playback/test3.c: (update_scale), (main):
- More playbin fixes. Added README. Do better element filtering.
- Added base class to preroll media. Added test apps.
+2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ tests/check/Makefile.am: See if this makes the build bots happy.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ See if this makes the build bots happy.
+ * tests/check/libs/cddabasesrc.c:
+ UTF8-ise my name.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_flush_decoder):
- * ext/mpeg2dec/gstmpeg2dec.h:
- various debugging improvements. Reset stream to next picture
- instead of sequence header, otherwise seeks cannot work.
+2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst/subparse/samiparse.c: (handle_start_font),
+ (fix_invalid_entities):
+ More case-insensitivity for certain tags; recognise entities with
+ decimal codes as special entities as well (#357330).
- * gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type),
- (gst_video_box_class_init), (gst_video_box_set_property),
- (gst_video_box_i420), (gst_video_box_ayuv), (gst_video_box_chain):
- Use pad_alloc where possible.
+2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
+ Original commit message from CVS:
+ * gst-libs/gst/Makefile.am:
+ Need to build tag directory before cdda.
- * sys/oss/gstosselement.c: (gst_osselement_reset),
- (gst_osselement_parse_caps):
- * sys/oss/gstosselement.h:
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Fix offset on osssrc.
+2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_base_init):
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
+ (gst_tag_register_musicbrainz_tags):
+ Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
+ depend on libgsttag. This is required so we can extract/read tags like
+ DISCID without depending on libgstcddabasesrc (which used to register
+ them).
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
+ tags (also see #347848).
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
+ Log vorbis comments we are actually writing. Const-ify array.
-2004-07-07 Wim Taymans <wim@fluendo.com>
+2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/theora/theora.c: (plugin_init):
- * ext/theora/theoradec.c: (theora_dec_from_granulepos),
- (theora_dec_src_query), (theora_dec_chain):
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_buffer_from_packet),
- (theora_push_packet), (theora_enc_chain):
- Fix theora granulepos calculation.
- Fix overflow in duration/position calculation.
- Bump rank to PRIMARY for theoradec.
- Use granulepos of last packet to calculate position.
- Set keyframe flag on buffers when needed.
+ gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gen_preroll_element):
+ Improve buffering a bit by avoiding a deadlock because we cannot assume
+ the underrun is always called.
-2004-07-06 David Schleef <ds@schleef.org>
+2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
- * gst/playback/Makefile.am: 'test' in bin_PROGRAMS? Are you
- serious? (Fixed, obviously.)
+ gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Added MPEG-4 AAC and id and caps. Fixes #357289
+ Added WMA9 Lossless id.
-2004-07-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/LINGUAS:
- * po/cs.po:
- added Czech translation (Miloslav Trmac)
-
-2004-07-05 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (gst_decode_bin_init), (gst_decode_bin_dispose),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (close_link), (type_found), (gst_decode_bin_set_property),
- (gst_decode_bin_get_property), (gst_decode_bin_get_event_masks),
- (gst_decode_bin_send_event), (gst_decode_bin_get_formats),
- (gst_decode_bin_convert), (gst_decode_bin_get_query_types),
- (gst_decode_bin_query), (plugin_init):
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (rebuild_pipeline), (get_audio_element),
- (get_video_element), (new_pad), (setup_source),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (gst_play_bin_change_state), (gst_play_bin_add_element),
- (gst_play_bin_remove_element), (gst_play_bin_get_event_masks),
- (gst_play_bin_send_event), (gst_play_bin_get_formats),
- (gst_play_bin_convert), (gst_play_bin_get_query_types),
- (gst_play_bin_query), (gst_play_bin_get_clock), (plugin_init):
- * gst/playback/test.c: (main):
- More fixes, cleaned up playbin, make it use decodebin. Added
- threaded property to playbin.
-
-2004-07-05 Wim Taymans <wim@fluendo.com>
+ ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Fix misleading docs addition.
+ * tests/check/elements/videotestsrc.c: (check_rgb_buf):
+ Get rid of compiler warning the right way.
- * configure.ac:
- * gst/playback/Makefile.am:
- * gst/playback/decodetest.c: (main):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (gst_decode_bin_init), (gst_decode_bin_dispose),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (close_link), (type_found), (gst_decode_bin_set_property),
- (gst_decode_bin_get_property), (gst_decode_bin_change_state),
- (gst_decode_bin_get_event_masks), (gst_decode_bin_send_event),
- (gst_decode_bin_get_formats), (gst_decode_bin_convert),
- (gst_decode_bin_get_query_types), (gst_decode_bin_query),
- (plugin_init):
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (gen_default_output), (rebuild_pipeline),
- (collect_sink_pads), (find_compatibles), (close_pad_link),
- (try_to_link_1), (new_pad), (close_link), (type_found),
- (setup_source), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gst_play_bin_factory_filter),
- (compare_ranks), (gst_play_bin_collect_factories),
- (gst_play_bin_change_state), (gst_play_bin_add_element),
- (gst_play_bin_remove_element), (gst_play_bin_get_event_masks),
- (gst_play_bin_send_event), (gst_play_bin_get_formats),
- (gst_play_bin_convert), (gst_play_bin_get_query_types),
- (gst_play_bin_query), (gst_play_bin_get_clock), (plugin_init):
- * gst/playback/test.c: (main):
- Added some playback helper elements and some test apps, very alpha
- still.
-
-2004-07-04 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_xrun_recovery):
- only restart audio when we indeed have an xrun to fix repeated
- xruns. Fix suggested by Giuliano Pochini.
-
-2004-07-03 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper): Disable
- call to gst_debug_log() if debugging is disabled (bug #145118)
-
-2004-07-03 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_xrun_recovery):
- use our own functions for restarting the alsa device.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- I should apply patches myself - use MIN for the third argument, not
- the second, this fixes seeking
-
-2004-07-02 David Schleef <ds@schleef.org>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_class_init),
- (gst_flacdec_write): Actually, GST_PAD_CAPS() has nothing to
- do with the logic.
-
-2004-07-02 David Schleef <ds@schleef.org>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_write): Set duration on
- output buffers. Fix logic mistake. (bug #144866)
-
-2004-07-02 David Schleef <ds@schleef.org>
-
- * gst-libs/gst/xoverlay/Makefile.am: xoverlay no longer depends
- on X. (bug #144753)
-
-2004-07-02 David Schleef <ds@schleef.org>
-
- * gst/wavenc/gstwavenc.c: (gst_wavenc_setup),
- (gst_wavenc_stop_file): Switch to GST_WRITE_UINT32_LE macros
- (bug #144624)
- * sys/oss/gstosselement.c: (gst_osselement_probe_caps),
- (gst_osselement_rate_probe_check): Add another workaround for
- buggy drivers (bug #145336)
-
-2004-07-02 David Schleef <ds@schleef.org>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_handle_client_write):
- Most systems don't have MSG_NOSIGNAL.
-
-2004-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_finalize),
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_push_full),
+ (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
+ (gst_base_rtp_depayload_process),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_queue_release):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Small cleanups.
+ Fix some leaks.
+ Refactored the process method and added methods to push from the process
+ vmethod.
+ Use _scale functions.
+ API: gst_base_rtp_depayload_push_ts
+ API: gst_base_rtp_depayload_push
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ timestamps are uint.
+
+2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ Remove unused statement from doc example.
- * Makefile.am:
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- (hopefully) fix both install and dist and make error message useful.
- needs testing across automakes.
+2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-02 Benjamin Otte <otte@gnome.org>
+ * gst/videorate/gstvideorate.c:
+ update docs
+ Original commit message from CVS:
+ update docs
+
+2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/videoorientation.c:
+ (gst_video_orientation_iface_init),
+ (gst_video_orientation_get_hflip),
+ (gst_video_orientation_get_vflip),
+ (gst_video_orientation_get_hcenter),
+ (gst_video_orientation_get_vcenter),
+ (gst_video_orientation_set_hflip),
+ (gst_video_orientation_set_vflip),
+ (gst_video_orientation_set_hcenter),
+ (gst_video_orientation_set_vcenter):
+ Add since tags to new API docs, ChangeLog surgery (forgot API keyword
+ in ChangeLog)
- * ext/ogg/gstogg.c: (plugin_init):
- we require bytestream now
- * ext/ogg/gstoggdemux.c:
- huge diff to implement chain setup in a fast and generic way. This
- improves tag reading and startup of huge files (read: Theora videos)
- quite a bit. It probably contains bugs, too, so please test.
- Seeking is not improved to the fast method.
+2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-29 Wim Taymans <wim@fluendo.com>
+ tests/check/: but disable for now since it doesn't pass (something wrong with
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
+ (create_rgb_conversions), (rgb_conversion_free),
+ (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
+ (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
+ Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
+ but disable for now since it doesn't pass (something wrong with
+ RGBA somewhere).
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- * ext/ogg/gstoggmux.c:
- Fix memleak in oggdemux when running unconnected pads.
- doc update in mux, start working on keyframe mode.
-
-2004-06-29 Benjamin Otte <otte@gnome.org>
-
- * sys/oss/gstosssink.c:
- * sys/oss/gstosssrc.c:
- advertise correct template caps - we indeed do non-native endianness
- and 8bit audio has no endianness
- * sys/ximage/ximagesink.c: (gst_ximagesink_getcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_getcaps):
- avoid (wrong) duplications in getcaps function and return
- template caps
-
-2004-06-29 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_multifdsink_class_init), (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_clear),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_thread),
- (gst_multifdsink_init_send), (gst_multifdsink_close):
- Fix wrong GList iteration that could crash the server when
- more then 2 clients disconnect at the same time. Read all the
- pending commands in one batch to recover from command storms under
- very heavy load.
-
-2004-06-28 Wim Taymans <wim@fluendo.com>
-
- * gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
- (gst_videomixer_pad_class_init), (gst_videomixer_pad_get_property),
- (gst_videomixer_pad_set_property),
- (gst_videomixer_pad_sinkconnect), (gst_videomixer_pad_init),
- (gst_video_mixer_background_get_type), (gst_videomixer_get_type),
- (gst_videomixer_class_init), (gst_videomixer_init),
- (gst_videomixer_request_new_pad), (gst_videomixer_blend_ayuv_i420),
- (pad_zorder_compare), (gst_videomixer_sort_pads),
- (gst_videomixer_fill_checker), (gst_videomixer_fill_color),
- (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers),
- (gst_videomixer_update_queues), (gst_videomixer_loop),
- (plugin_init):
- Avoid divide by zero, choose masterpad as the pad with the highest
- framerate.
-
-2004-06-27 Julien Moutte <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate),
- (gst_ximagesink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
- (gst_xvimagesink_xwindow_new): I prefer locking the mutex in the
- function directly. We might want to call it from somewhere else one day.
-
-2004-06-27 Julien Moutte <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate),
- (gst_ximagesink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
- (gst_xvimagesink_xwindow_new): Trying to fix the random behaviour of
- window decorations.
-
-2004-06-27 Wim Taymans <wim@fluendo.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_class_init), (gst_dvdec_init),
- (gst_dvdec_video_getcaps), (gst_dvdec_video_link),
- (gst_dvdec_push), (gst_dvdec_loop), (gst_dvdec_change_state),
- (gst_dvdec_set_property), (gst_dvdec_get_property):
- * ext/dv/gstdvdec.h:
- Implement drop_factor property to lower the framerate with
- a factor.
-
-2004-06-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- unbreak Company's fix that didn't install the -enum.h files
-
-2004-06-27 Wim Taymans <wim@fluendo.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_push), (gst_dvdec_loop),
- (gst_dvdec_change_state):
- * ext/dv/gstdvdec.h:
- Fix timestamp, duration and offset of the buffers.
-
-2004-06-27 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_multifdsink_class_init), (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_clear),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_thread),
- (gst_multifdsink_init_send), (gst_multifdsink_close):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpserversink.c:
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_handle_select), (gst_tcpserversink_close):
- More multifdsink fixes, more recovery policy fixes.
- Removed stupid g_print
+2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
+ (queue_out_of_data), (gen_preroll_element),
+ (preroll_remove_overrun), (probe_triggered):
+ Refactor handling of overrun detection.
+ Separate handling of group completion and deadlock detection when doing
+ network buffering. This should fix some deadlocks that were not detected
+ because the group was completed.
+ Add more comments, improve debugging.
+
+2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/: Some more compilation fixes.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/libs/audio.c:
+ Some more compilation fixes.
+
+2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ Early morning compilation fix.
+
+2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * configure.ac:
+ bump nano
+ Original commit message from CVS:
+ bump nano
+
+2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/: Fix some warnings.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/elements/multifdsink.c: (GST_START_TEST):
+ * tests/check/elements/videorate.c: (GST_START_TEST):
+ * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
+ * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
+ Fix some warnings.
+
+2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_get_times):
+ change colorkey behaviour back according to #354773 comment 6/7
-2004-06-26 Wim Taymans <wim@fluendo.com>
+2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/Makefile.am:
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_multifdsink_get_type), (gst_multifdsink_base_init),
- (gst_multifdsink_class_init), (gst_multifdsink_init),
- (gst_multifdsink_debug_fdset), (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_thread),
- (gst_multifdsink_chain), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property), (gst_multifdsink_init_send),
- (gst_multifdsink_close), (gst_multifdsink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpplugin.c: (plugin_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_get_type),
- (gst_tcpserversink_class_init), (gst_tcpserversink_init),
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_handle_select),
- (gst_tcpserversink_set_property), (gst_tcpserversink_get_property),
- (gst_tcpserversink_init_send), (gst_tcpserversink_close):
- * gst/tcp/gsttcpserversink.h:
- Added multifdsink, made tcpserversink a subclass of fdsink, removed
- one of the locks, added recovery policy to multifdsink.
+ * ChangeLog:
+ ChangeLog surgery: remove junk
+ Original commit message from CVS:
+ ChangeLog surgery: remove junk
+
+2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
+ (gst_multi_fd_sink_recover_client),
+ (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
+ (gst_multi_fd_sink_get_property):
+ * gst/tcp/gstmultifdsink.h:
+ Implement stubbed out properties unit-type, units-soft-max,
+ units-max, to allow specifying maximum sizes in units other than
+ buffers.
+ Fixes #355935
-2004-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Reorder the audio formats a bit for clarity.
+ Detect and create caps for MSGSM and MSN (WAV49).
+ Fixes #356596.
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
+ Small cleanups, move error handling out of normal flow for clarity.
+
+2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add new interface to control video orientation (fixes #354908)
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/videoorientation.c:
+ (gst_video_orientation_get_type),
+ (gst_video_orientation_iface_init),
+ (gst_video_orientation_get_hflip),
+ (gst_video_orientation_get_vflip),
+ (gst_video_orientation_get_hcenter),
+ (gst_video_orientation_get_vcenter),
+ (gst_video_orientation_set_hflip),
+ (gst_video_orientation_set_vflip),
+ (gst_video_orientation_set_hcenter),
+ (gst_video_orientation_set_vcenter):
+ * gst-libs/gst/interfaces/videoorientation.h:
+ Add new interface to control video orientation (fixes #354908)
+
+2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ Use G_UNLIKELY in _create and log one more detail.
+ (gst_video_test_src_get_times), (gst_video_test_src_create):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
+ Use gst_util_uint64_scale_int in _get_times().
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- fix decision for when getting frames with same timestamp
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
- (gst_v4lsrc_get), (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- * sys/v4l/gstv4lsrc.h:
- add latency offset property
+2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
+ Give better warning message (add object and detail).
- * gst/videorate/gstvideorate.c: (gst_videorate_chain),
- (plugin_init):
- fix debugging. add category.
+2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_get_times):
+ xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
+ #354773), use gst_util_uint64_scale_int in _get_times()
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- fix wrong offsets
+2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
+ Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
+ always true, leading to dropping all timestamps.
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
- (gst_alsa_src_get_time), (gst_alsa_src_loop),
- (gst_alsa_src_change_state):
- return a time that is in sync with the element's processing
+2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-06-25 Wim Taymans <wim@fluendo.com>
+ ext/libvisual/visual.c: update to work also with libvisual 0.4 API
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_vis_src_negotiate),
+ (gst_visual_chain), (gst_visual_change_state):
+ update to work also with libvisual 0.4 API
+ * tools/gst-launch-ext.1.in:
+ * tools/gst-visualise.1.in:
+ remove references to old man-pages
+ * tests/examples/seek/seek.c: (main):
+ add real meadi-buttons, add tool-tips for the seek-options, arrange
+ seek options in a table
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_init), (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_client_remove),
- (gst_tcpserversink_handle_client_read),
- (gst_tcpserversink_client_queue_data),
- (gst_tcpserversink_client_queue_caps),
- (gst_tcpserversink_client_queue_buffer),
- (gst_tcpserversink_handle_client_write),
- (gst_tcpserversink_queue_buffer),
- (gst_tcpserversink_handle_clients), (gst_tcpserversink_thread),
- (gst_tcpserversink_chain), (gst_tcpserversink_set_property),
- (gst_tcpserversink_get_property), (gst_tcpserversink_init_send),
- (gst_tcpserversink_close):
- * gst/tcp/gsttcpserversink.h:
- Serversink rewrite. Really do non blocking writes to clients and
- maintain an internal queue to handle slower clients while not
- disturbing fast clients.
+2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
+ (gst_ogg_mux_push_buffer):
+ Don't generate out-of-order timestamps from oggmux, instead clamp
+ output timestamps to be >= the previously output ts.
+ Fixes #355595
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- better debug, don't override OFFSET and OFFSET_END
+2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
-2004-06-25 Iain <iain@prettypeople.org>
+ gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_multi_fd_sink_class_init):
+ Updates, fixes, and typo corrections for multifdsink. No functional
+ changes.
- * gst-libs/gst/media-info/media-info-priv.c (gmi_set_mime): Add
- name=source for the wavparse pipeline.
+2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
-2004-06-24 Johan Dahlin <johan@gnome.org>
+ gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
+ Don't crash on truncated files - check that we got an 8 byte buffer
+ before trying to memcmp it.
- * ext/theora/theoraenc.c (theora_enc_chain): Call
- gst_pad_try_set_caps instead of gst_pad_set_explicit_caps so the
- streamheader caps are set correctly.
+2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (get_active_source):
+ Make stream-switching appear instant to the application
+ (ie. make sure that a g_object_get on 'current-foo' returns
+ the stream previously set with g_object_set(). Totem needs
+ this to update stream-related meta-info (like audio-codec)
+ correctly when switching streams.
- * ext/vorbis/vorbisenc.c: (raw_caps_factory),
- (gst_vorbisenc_setup), (gst_vorbisenc_set_property):
- respect minimum bitrate; same could be done for max bitrate
+2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
+ (gst_alsa_mixer_ensure_track_list):
+ Try harder to guess which mixer track is the master mixer
+ track (instead of just taking the first one that has a pvolume).
+ Fixes #342228.
- * ext/vorbis/vorbisenc.c: (raw_caps_factory),
- (gst_vorbisenc_setup):
- fix sample rate range
+2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
+ (gst_audio_convert_transform_caps):
+ Get structure-name just once.
+
+2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/: Fix big batch of compiler warnings.
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c: (GST_START_TEST):
+ * tests/check/elements/videotestsrc.c: (check_rgb_buf):
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ * tests/check/elements/vorbisdec.c: (GST_START_TEST):
+ * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
+ (test_pipeline), (GST_START_TEST):
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
+ Fix big batch of compiler warnings.
+
+2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Add docs about icydemux usage in connection with gnomevfssrc
+ * ext/libvisual/visual.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst/audiorate/gstaudiorate.c:
+ More G_OBJECT macro fixing.
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Fix wrong info in header due to copy & paste
+
+2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
+ (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
+ (gst_base_audio_src_create), (gst_base_audio_src_change_state):
+ Do the delay calculation in the source/sink base classes as this is
+ specific for the capture/playback mode.
+ Try to fixate a bit better, like round depth up to a multiple of 8
+ bigger than width.
+ Handle underruns correctly by marking DISCONT on buffers and adjusting
+ timestamps to handle the gap.
+ Set offset/offset_end correctly on buffers.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
+ (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ Remove resync and underrun recovery from the ringbuffer.
+ Fix ringbuffer read code on under/overrun.
+
+2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_init), (fill_buffer), (check_queue),
+ (queue_threshold_reached), (gst_play_base_bin_set_property),
+ (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybasebin.h:
+ Don't use a 0 low watermark when buffering, it is catching starvation
+ way too late. Instead, use a 3 second queue with 30 and 95
+ percent low/high watermarks.
+ Added queue-min-threshold property to configure low watermark.
+ Use new _buffering message API.
+ Make queue_threshold variable big enough to store a uint64 time value.
+ API: playbin::queue-min-threshold property.
+
+2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ configure.ac: We require 0.10.10.1 now because of _wait_preroll().
+ Original commit message from CVS:
+ * configure.ac:
+ We require 0.10.10.1 now because of _wait_preroll().
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Use gst_base_sink_wait_preroll().
+
+2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/: Use DEBUG_OBJECT more.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
+ * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
+ Use DEBUG_OBJECT more.
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
+=== release 0.10.10 ===
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_class_init),
- (gst_oggvorbisenc_setup):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init),
- (gst_vorbisenc_setup):
- resolve ambiguities in code and description
+2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-24 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/theora/theoraparse.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst/playback/gstplaybin.c:
+ * tests/check/Makefile.am:
+ * win32/common/config.h:
+ releasing 0.10.10
+ Original commit message from CVS:
+ releasing 0.10.10
- * ext/alsa/gstalsa.c: (gst_alsa_start), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
- (gst_alsa_src_update_avail), (gst_alsa_src_loop):
- Use alsa trigger_tstamp to get the timestamp of the first
- sample in the buffer for more precise sync. Some cleanups.
+2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-24 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * win32/common/config.h:
+ second prerelease
+ Original commit message from CVS:
+ second prerelease
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_link),
- (gst_audiorate_init), (gst_audiorate_chain),
- (gst_audiorate_set_property), (gst_audiorate_get_property):
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_chain):
- Added some logging, fixed an overflow bug in videorate.
+2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-24 Benjamin Otte <otte@gnome.org>
+ * ChangeLog:
+ update bug in changelog
+ Original commit message from CVS:
+ update bug in changelog
- * ext/kio/Makefile.am:
- fix for builddir != srcdir and distcheck
+2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
-2004-06-24 Benjamin Otte <otte@gnome.org>
+ Fix implementation of sync-method 'next-keyframe'
+ Original commit message from CVS:
+ patch by: Michael Smith <msmith at fluendo dot com>
+ * gst/tcp/gstmultifdsink.c: (is_sync_frame),
+ (gst_multi_fd_sink_client_queue_buffer),
+ (gst_multi_fd_sink_new_client):
+ * tests/check/elements/multifdsink.c: (GST_START_TEST),
+ (multifdsink_suite):
+ Fix implementation of sync-method 'next-keyframe'
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- * gst/tcp/Makefile.am:
- * sys/dxr3/Makefile.am:
- don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
- tarball.
- Also add all *.list files that were missing.
- * Makefile.am:
- add a distcheck hook to ensure the above doesn't happen again.
+2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
-2004-06-23 David I. Lehn <dlehn@users.sourceforge.net>
+ ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
+ Original commit message from CVS:
+ patch by: Wim Taymans <wim at fluendo dot com>
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
+ This patch removes the RANDOM flag that was incorrectly introduced with
+ revision 1.91. Fixes #354590
- * ext/Makefile.am: s/DTS_DIR=dvdread/DTS_DIR=dts/
+2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-23 Colin Walters <walters@redhat.com>
+ * configure.ac:
+ * win32/common/config.h:
+ first prerelease
+ Original commit message from CVS:
+ first prerelease
- * m4/Makefile.am: Distribute gst-fionread.m4.
+2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ update po files
+ Original commit message from CVS:
+ update po files
- * configure.ac: back to dev
+2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-23 Wim Taymans <wim@fluendo.com>
+ tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Random variation in Makefile line to see if it makes the
+ gen64-base-full bot any happier.
- * ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
- (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
- (gst_alsa_sink_loop), (gst_alsa_sink_get_time):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_init),
- (gst_alsa_src_get_time), (gst_alsa_src_update_avail),
- (gst_alsa_src_loop):
- Add clock to alsasrc. Take new capture timestamp when
- restarting after an overrun. Split up some functions between
- alsasrc and alsasink.
-
-=== release 0.8.2 ===
-
-2004-06-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_dispose),
- (gst_alsa_change_state), (gst_alsa_update_avail),
- (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- merge back changes from release
+2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-23 Wim Taymans <wim@fluendo.com>
+ tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c: (oggmux_suite):
+ Disable test that fails at the moment (killed after timeout).
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_class_init),
- (gst_audiorate_init), (gst_audiorate_chain),
- (gst_audiorate_set_property), (gst_audiorate_get_property):
- Implement sample dropping and notify
+2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
+
+ tests/check/: Add simple unit test for oggmux from #337026 with checking for the
+ Original commit message from CVS:
+ Patch by: James Livingston <doclivingston at gmail.com>
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/.cvsignore:
+ * tests/check/pipelines/oggmux.c: (get_page_codec),
+ (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
+ (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
+ (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
+ (test_theora_vorbis), (oggmux_suite):
+ Add simple unit test for oggmux from #337026 with checking for the
+ EOS flags disabled for the time being.
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_buffer_from_packet),
- (theora_push_packet), (theora_enc_chain):
- Some cleanups, make sure the timestamps are correct.
+ ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
+ Original commit message from CVS:
+ patch by: Alessandro Dessina <alessandro nnva org>
+ * ext/ogg/gstoggmux.c:
+ Add cmml caps to oggmux. Fixes #353912
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsa.c: (gst_alsa_get_time), (gst_alsa_clock_update),
- (gst_alsa_change_state), (gst_alsa_update_avail),
- (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- Cleanups, take queued samples into account when reporting
- the time.
+ tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
+ Original commit message from CVS:
+ * tests/check/elements/videotestsrc.c: (check_rgb_buf):
+ Returning a return value often helps. In this case, we
+ don't need the return value anyway, so just get rid of it.
+ Should make build bots much happier.
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
+ (paint_get_structure), (gst_video_test_src_get_size),
+ (gst_video_test_src_smpte), (gst_video_test_src_snow),
+ (gst_video_test_src_unicolor), (paint_setup_AYUV),
+ (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
+ (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
+ * gst/videotestsrc/videotestsrc.h:
+ Add support for AYUV and the various RGBA formats. Initialise
+ fields of paintinfo structs allocated on the stack.
+ * tests/check/elements/videotestsrc.c: (right_shift_colour),
+ (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
+ (GST_START_TEST), (videotestsrc_suite):
+ Add unit tests for videotestsrc's RGB output.
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_init):
- Initialize the property as well.
+2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_pattern_get_type),
+ (gst_video_test_src_set_pattern):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
+ (gst_video_test_src_black), (gst_video_test_src_white),
+ (gst_video_test_src_red), (gst_video_test_src_green),
+ (gst_video_test_src_blue):
+ * gst/videotestsrc/videotestsrc.h:
+ Add more uni-colour patterns ("white", "red", "green", and "blue").
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_set_property), (gst_videorate_get_property):
- Add property to make videorate silent.
- Add property to prefer new frames over old ones.
+ gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
+ Fix stride for YVYU, should be word-aligned (#353658).
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/osxvideo/Makefile.am:
- Workaround so that the osxvideo .so file gets linked with the
- Cocoa, OpenGL and QuickTime frameworks
+ gst/adder/gstadder.c: Fix build.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_src_event):
+ Fix build.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (forward_event_func),
+ (gst_adder_src_event), (gst_adder_collected),
+ (gst_adder_change_state):
+ * gst/adder/gstadder.h:
+ Remember the start position asked in the incoming seeks, so we can
+ output GST_EVENT_NEW_SEGMENT with a correct position value (instead
+ of assuming it will always be 0).
+
+2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
+ (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_loop):
+ Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
+
+2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_get_unit_size):
+ Return FALSE instead of returning a random false unit
+ size when the format isn't known/supported (even if
+ this shouldn't happen under normal circumstances).
- * sys/osxaudio/Makefile.am:
- Workaround so that the osxaudio .so file gets linked with the
- CoreAudio framework
+2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
+ Original commit message from CVS:
+ Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
+ (gst_gnome_vfs_src_start):
+ Try harder to get the size from a uri by using _info_uri() when
+ _info_from_handle() does not give us enough info.
+ Also follow symlinks when getting the size.
+ Partially Fixes #332864.
+
+2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
+
+ ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
+ Original commit message from CVS:
+ Patch by: Viktor Peters <viktor dot peters at gmail dot com>
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
+ (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
+ (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
+ (gst_alsa_mixer_set_record):
+ * ext/alsa/gstalsamixertrack.c:
+ (gst_alsa_mixer_track_update_alsa_capabilities),
+ (alsa_track_has_cap), (gst_alsa_mixer_track_new),
+ (gst_alsa_mixer_track_update):
+ * ext/alsa/gstalsamixertrack.h:
+ Improve and fix mixer track handling, in particular better handling
+ of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
+ track objects for tracks that have both capture and playback volume
+ (and label them differently as well so they're not mistakenly
+ assumed to be duplicates); classify mixer tracks that only affect
+ the audible volume of something (rather than the capture volume)
+ as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
+ for capture tracks to correspond to alsa-pswitch alsa-cswitch
+ (following the meaning documented in the mixer interface header
+ file); add support for alsa's exclusive cswitch groups; update/sync
+ state/flags better if mixer settings are changed by another
+ application. Fixes #336075.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Whoops, my fault...fixed build issues
+ gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ Improve docs: add section about BUFFERING messages sent by playbin.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- Add objective-c support if running in Darwin/Mac OS X
- * sys/Makefile.am:
- * sys/osxvideo:
- * sys/osxvideo/Makefile.am:
- * sys/osxvideo/osxvideosink.h:
- * sys/osxvideo/osxvideosink.m:
- * sys/osxvideo/cocoawindow.h:
- * sys/osxvideo/cocoawindow.m:
- Add osxvideosink, a cocoa-based osx video sink
-
-
-2004-06-19 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/dvdnav/gst-dvd:
- Grab the gconf key from the right spot
- * gst/debug/gstnavseek.c: (gst_navseek_init),
- (gst_navseek_segseek), (gst_navseek_handle_src_event),
- (gst_navseek_chain):
- * gst/debug/gstnavseek.h:
- Add 's', 'e' and 'l' keypresses to navseek to define the start,end
- and loop parameters of a segment seek.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
- (gst_videotestsrc_get_event_masks),
- (gst_videotestsrc_handle_src_event), (gst_videotestsrc_get):
- * gst/videotestsrc/gstvideotestsrc.h:
- Add seeking support to videotestsrc
- Initialise the timestamp_offset variable.
+ ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
+ (gst_vorbis_enc_buffer_check_discontinuous),
+ (gst_vorbis_enc_chain):
+ Ignore explicit DISCONT marked on buffers (which is often spurious,
+ particularly when using multiple segments), in favour of solely
+ using the timestamps/durations.
-2004-06-18 Wim Taymans <wim@fluendo.com>
+2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/sidplay/gstsiddec.cc:
- Fix negotiation and set correct end offset.
+ gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Don't rely on incoming buffers offset anymore, since it is completely
+ broken when using multiple segments.
+ Instead convert the incoming buffers timestamp to running time, and
+ then convert that value to the offsets.
+ Also inform GstSegment of the last outputted stop position, which is
+ needed if we received several segments with an unknown stop value.
-2004-06-18 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: branch and prerelease
+ ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
+ fix buffer unreffing on a header push failure
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init),
- (gst_tcpclientsrc_getcaps), (gst_tcpclientsrc_get),
- (gst_tcpclientsrc_init_receive):
- * gst/tcp/gsttcpclientsrc.h:
- read caps when connecting to server for GDP so we set them correctly
-
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- notify drops and duplicates
- * gst/videoscale/videoscale.c: (videoscale_get_structure):
- no good reason to limit ourselves to 100x100
-
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
- (gst_v4lsrc_open), (gst_v4lsrc_src_link), (gst_v4lsrc_getcaps),
- (gst_v4lsrc_get), (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4l_calls.c: (gst_v4l_set_window_properties),
- (gst_v4l_open), (gst_v4l_get_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_grab_frame),
- (gst_v4lsrc_try_capture):
- * sys/v4l/v4lsrc_calls.h:
- change try_palette to more general try_capture
- add autoprobe option so we can turn off autoprobing
- various fixes
+ gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
+ (gst_audio_rate_chain):
+ Make the metadata of the buffer writable before changing its
+ flags.
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- add videorate
- * sys/ximage/ximagesink.c: (gst_ximagesink_finalize),
- (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_finalize),
- (gst_xvimagesink_class_init):
- run them as finalize, not dispose, since dispose can be invoked
- multiple times
-
-2004-06-17 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_dispose),
- (gst_alsa_get_time), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_get_type):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_init), (gst_alsa_src_loop),
- (gst_alsa_src_change_state):
- * ext/alsa/gstalsasrc.h:
- Make the xrun code timestamp and offset the buffers correctly.
- moved the clock to the base class, use alsa methods to get time.
- Do correct timestamping on outgoing buffers.
+ * ChangeLog:
+ Fix changelog with bugzilla bug it fixed.
+ Original commit message from CVS:
+ Fix changelog with bugzilla bug it fixed.
+
+2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiorate/gstaudiorate.c: Fix audiorate some more.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
+ (gst_audio_rate_setcaps), (gst_audio_rate_init),
+ (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
+ (gst_audio_rate_chain), (gst_audio_rate_change_state):
+ Fix audiorate some more.
+ Reset and resync counters on flush and READY.
+ Handle the DISCONT flag correctly.
+ Use GstSegment to track position.
+ Fail when not negotiated.
+
+2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/tcp/gstmultifdsink.c: Fix spelling.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
+ Fix spelling.
+ Remove accidently included debug line.
-2004-06-17 Wim Taymans <wim@fluendo.com>
+2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/audiorate/Makefile.am:
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_get_type),
- (gst_audiorate_base_init), (gst_audiorate_class_init),
- (gst_audiorate_link), (gst_audiorate_init), (gst_audiorate_chain),
- (gst_audiorate_set_property), (gst_audiorate_get_property),
- (gst_audiorate_change_state), (plugin_init):
- Added an audiorate converter that fills in gaps.
-
-2004-06-17 Johan Dahlin <johan@gnome.org>
-
- * ext/tcp/*: Revert Zaheer changes, to make things actually work again.
-
-2004-06-16 Wim Taymans <wim@fluendo.com>
-
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_sync_mode_get_type),
- (gst_v4lsrc_class_init), (gst_v4lsrc_init), (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get), (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- * sys/v4l/gstv4lsrc.h:
- Added a copy mode to v4lsrc where it will output a copied version
- of its internal hardware buffer.
- Fix the wrong FLAG_SET usage. The flags are integers, not bits, you
- can't | them.
-
-2004-06-16 Wim Taymans <wim@fluendo.com>
-
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Timestamp fixes.
-
-2004-06-16 Wim Taymans <wim@fluendo.com>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_sync_mode_get_type),
- (gst_v4lsrc_class_init), (gst_v4lsrc_init), (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get), (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- Added a sync mode enum property to control v4lsrc timestamp method
- Removed the use-fixed-fps property and moved functionality in
- the enum.
- Don't error on an error value from v4l-conf, it might not always
- be a real error.
-
-2004-06-16 Wim Taymans <wim@fluendo.com>
+ gst/tcp/gstmultifdsink.c: Small cleanups.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
+ Small cleanups.
+ If a buffer is received with no caps, make the buffer metadata
+ writable and set the caps, making sure that we don't screw up the
+ refcounts.
- * gst/videorate/Makefile.am:
- * gst/videorate/gstvideorate.c: (gst_videorate_get_type),
- (gst_videorate_base_init), (gst_videorate_class_init),
- (gst_videorate_getcaps), (gst_videorate_link),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_set_property), (gst_videorate_get_property),
- (gst_videorate_change_state), (plugin_init):
- Added a video timestamp corrector.
+2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
+ (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
+ Fix memory leaks and misleading debug messages, add a couple of
+ comments.
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
+ (gst_multi_fd_sink_render):
+ Do not use gst_buffer_make_writable() in a basesink render method,
+ as it may incorrectly unref the buffer. Instead, use convoluted
+ dance to avoid copying the buffer except when we need to.
- fixed a potential leak with previous commit
+2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
+ ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c:
+ (gst_vorbis_enc_buffer_check_discontinuous):
+ Allow very small discontinuities in the timestamps. These we can't
+ do anything useful with anyway (because vorbis's timestamps have
+ only sample granularity), and are commonly produced by elements with
+ minor bugs. Allow up to 1/2 a sample out.
+ Fixes #351742.
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
- Added missing refcount, fixes bug #144425
- Cheers Tim for finding the bug
+ tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
+ (play_scrub_toggle_cb), (main):
+ Add a checkbox to enable play scrubbing. Makes it possible to disable
+ normal scrubbing.
-2004-06-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/gstv4lcolorbalance.c:
- * sys/v4l/gstv4lcolorbalance.h:
- * sys/v4l/gstv4lelement.c:
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsink.h:
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4lmjpegsrc.h:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_fps_list),
- (gst_v4lsrc_get_fps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_get):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/gstv4ltuner.c:
- * sys/v4l/gstv4ltuner.h:
- * sys/v4l/gstv4lxoverlay.c:
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay),
- (gst_v4l_set_window), (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_set_window_properties),
- (gst_v4l_open), (gst_v4l_get_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_sync_frame),
- (gst_v4lmjpegsink_set_buffer), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start), (gst_v4lmjpegsink_get_buffer),
- (gst_v4lmjpegsink_play_frame), (gst_v4lmjpegsink_wait_frame),
- (gst_v4lmjpegsink_playback_stop),
- (gst_v4lmjpegsink_playback_deinit):
- * sys/v4l/v4lmjpegsink_calls.h:
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame),
- (gst_v4lmjpegsrc_sync_next_frame), (gst_v4lmjpegsrc_set_buffer),
- (gst_v4lmjpegsrc_set_capture), (gst_v4lmjpegsrc_set_capture_m),
- (gst_v4lmjpegsrc_capture_init), (gst_v4lmjpegsrc_capture_start),
- (gst_v4lmjpegsrc_grab_frame), (gst_v4lmjpegsrc_requeue_frame),
- (gst_v4lmjpegsrc_capture_stop), (gst_v4lmjpegsrc_capture_deinit):
- * sys/v4l/v4lmjpegsrc_calls.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_set_capture),
- (gst_v4lsrc_capture_init), (gst_v4lsrc_capture_start),
- (gst_v4lsrc_grab_frame), (gst_v4lsrc_requeue_frame),
- (gst_v4lsrc_capture_stop), (gst_v4lsrc_capture_deinit),
- (gst_v4lsrc_try_palette):
- * sys/v4l/v4lsrc_calls.h:
- bunch of paranoia cleanups
+ tests/check/elements/.cvsignore: make buildbot happy
+ Original commit message from CVS:
+ * tests/check/elements/.cvsignore:
+ make buildbot happy
-2004-06-14 David Schleef <ds@schleef.org>
+2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
+ (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
+ (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
+ (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
+ (gst_ogm_text_parse_strip_trailing_zeroes),
+ (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
+ (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
+ Refactor ogm parse, do better input checking, misc. clean-ups.
+ Cache incoming events and push them once the source pad has
+ been created. Don't pass unterminated strings to sscanf().
+ Strip trailing zeroes from subtitle text output, since they
+ are not valid UTF-8. Don't push vorbiscomment packets on
+ the subtitle text pad. Output perfect streams if possible.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_init),
- (cdparanoia_get), (cdparanoia_open), (cdparanoia_change_state):
- Send discont events and change timestamps appropriately when
- we get a seek event. (bug #144240)
- * ext/cdparanoia/gstcdparanoia.h:
+2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-06-14 Benjamin Otte <otte@gnome.org>
+ tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
+ Original commit message from CVS:
+ * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
+ Waits for tasks to settle down so that we clean up correctly for
+ valgrind.
- * ext/alsa/gstalsa.c: Use snd_pcm_hw_params_set_rate _near instead of
- snd_pcm_hw_params_set_rate since the latter fails for no good
- reason on some setups.
+2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-14 David Schleef <ds@schleef.org>
+ tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
+ Original commit message from CVS:
+ * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
+ Unit test fixes: \377 is more likely to fit into 8 bits than \777;
+ actually return return value in taglists_are_equal.
- * gst/volume/demo.c: (value_changed_callback): exp10() is not
- standard. Thank you for playing.
+2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-14 Wim Taymans <wim@fluendo.com>
+ ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
+ Fix crash due to broken bitstream parsing on x86-64: can't make
+ any assumptions about sizeof(struct) due to alignment/packing
+ differences on different architectures. Fixes #351790.
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
- Patch 1.3 broke the ordering of the colorspace info and
- made the plugin basically work by coincidence, reordered
- the info.
+2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
+ (gst_riff_parse_chunk), (gst_riff_parse_file_header),
+ (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
+ (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
+ (gst_riff_parse_info):
+ Protect public functions against bad input.
+ Do some cleanups.
+ Fix documentation.
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lame/gstlame.c:
- * ext/mad/gstmad.c:
- sync caps. Make sure mad can only output a list of rates, not
- a full range. In the future, have three caps lists for each of the
- mpeg versions. Change mpegversion to a double as well.
+ gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Add voxware audio IDs (even if we can't play it) (#351795).
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/volume/.cvsignore:
- * gst/volume/Makefile.am:
- * gst/volume/demo.c: (value_changed_callback), (idler),
- (setup_gui), (main):
- added small demo app
+ gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c:
+ (gst_riff_create_video_template_caps),
+ (gst_riff_create_audio_template_caps),
+ (gst_riff_create_iavs_template_caps):
+ Const-ify some arrays and use G_N_ELEMENTS instead
+ of wasting oodles of RAM on terminator bits.
-2004-06-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/esd/esdsink.c: (gst_esdsink_change_state):
- * ext/esd/esdsink.h:
- Close the esd connection on pause, because esd will just wait -
- blocking all other esd clients indefinitely.
+2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Christophe Fergeau <teuf@gnome.org>
+ And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ (gst_tag_list_to_vorbiscomment_buffer):
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ And the same for _to_vorbiscomment_buffer(): allow
+ id_data_len == 0 for speex.
- * gst/tags/gstvorbistag.c: replaced a g_warning which I added in my
- previous commit with GST_DEBUG
+2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/gdp/README:
+ adding a README
+ Original commit message from CVS:
+ adding a README
- * configure.ac:
- add a header check for a dvdread header in dvdnav. Fixes #133002
+2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Move GDP plugin to -base from -bad. Closes #347783.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * gst/gdp/Makefile.am:
+ * tests/check/Makefile.am:
+ Move GDP plugin to -base from -bad. Closes #347783.
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init_send):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init_receive):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init),
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_init_send):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
- * gst/tcp/gsttcpserversrc.h:
- Modified the tcp plugins so they are portable (IPv4,IPv6, any future
- version of IP)
+2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ (gst_tag_list_from_vorbiscomment_buffer):
+ Allow id_data_len == 0 (needed for vorbis comments in Speex files).
+ Also add some checks to make sure we don't memcmp() beyond the end of
+ vorbiscomment buffer if the ID to check for is larger than the buffer.
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Some more tests for gst_tag_list_from_vorbiscomment_buffer().
- * configure.ac:
- Added ogg library so that OSX detects libtheora properly
-
-2004-06-11 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_dec_chain),
- (theora_dec_change_state):
- Don't try to decode frames before we received a keyframe.
-
-2004-06-11 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_init), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property):
- Added property to set the maximum delay of a page.
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_init), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property):
- Added max-delay property to control the maximum amount
- of data to put in one page.
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (gst_theora_enc_init), (theora_enc_sink_link),
- (theora_buffer_from_packet), (theora_enc_set_property),
- (theora_enc_get_property):
- Set duration on encoded buffer, added some more properties
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- * ext/theora/theoraenc.c: (theora_enc_chain):
- Fix refcounting bugs
-
-2004-06-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
- (gst_asf_demux_loop), (gst_asf_demux_process_file),
- (gst_asf_demux_process_data), (gst_asf_demux_handle_data),
- (gst_asf_demux_process_object), (gst_asf_demux_get_stream),
- (gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event),
- (gst_asf_demux_handle_src_event), (gst_asf_demux_handle_src_query),
- (gst_asf_demux_change_state):
- * gst/asfdemux/gstasfdemux.h:
- You know Chimaira? "I - HATE - EVERYTHING". Yeah, that's what this
- feels like. I think we should set a new requirement for demuxers
- from now on to implement sane loop functions, data loops, query
- and seek functions before first commit into CVS. And this commit
- fixes all of the above.
-
-2004-06-10 Christophe Fergeau <teuf@gnome.org>
-
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add): make sure parsed
- vorbis comments are properly encoded in UTF-8 before adding them
- to a GstTagList
-
-2004-06-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (add_channels):
- handle min <= max correctly
- * ext/alsa/gstalsa.c: (gst_alsa_fixate_to_mimetype),
- (gst_alsa_fixate_field_nearest_int), (gst_alsa_fixate):
- add fixation functions so we fixate correctly. No preferring of alaw
- anymore because it's the first structure.
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsa.c: (gst_alsa_sw_params_dump),
- (gst_alsa_hw_params_dump):
- add functions to ease debugging in alsalib
- * ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
- (gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
- (gst_alsa_start_audio):
- only specify hw params if we really setup a format (fixes #134007 -
- or at least works around it)
-
-2004-06-09 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_push_page), (gst_ogg_mux_get_headers),
- (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_loop):
- Use stream caps to setup the initial pages in the ogg stream.
- Correctly set the streamheader caps on the srcpad.
-
-2004-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_fps_list),
- (gst_v4lsrc_get_fps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps):
- * sys/v4l/v4l_calls.c: (gst_v4l_set_window_properties),
- (gst_v4l_get_picture), (gst_v4l_get_audio), (gst_v4l_set_audio):
- add querying of fps lists for webcams. Negotiating to a framerate
- now works.
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_push_buffer), (theora_push_packet),
- (theora_set_header_on_caps), (theora_enc_chain):
- mark buffers and put on streamheader, raw theora streaming
- now works too, whee
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_header),
- (gst_tcp_gdp_read_caps):
- do a looping read for caps and GDP headers too
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get):
- return EOS instead of NULL in _get
-
-2004-06-08 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_header),
- (gst_tcp_gdp_read_caps), (gst_tcp_gdp_write_header),
- (gst_tcp_gdp_write_caps):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_gdp_read_caps),
- (gst_tcpserversrc_gdp_read_header), (gst_tcpserversrc_get):
- Memory leak fixes
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_base_init),
- (gst_vorbis_parse_class_init), (gst_vorbis_parse_init),
- (vorbis_parse_set_header_on_caps), (vorbis_parse_chain),
- (vorbis_parse_change_state):
- * ext/vorbis/vorbisparse.h:
- adding a vorbisparse element that marks the buffers, streaming
- raw vorbis using GDP now works, whee
-
-2004-06-08 Wim Taymans <wim@fluendo.com>
-
- * ext/jpeg/Makefile.am:
- * ext/jpeg/README:
- * ext/jpeg/gstjpeg.c: (plugin_init):
- * ext/jpeg/gstsmokedec.c: (gst_smokedec_get_type),
- (gst_smokedec_base_init), (gst_smokedec_class_init),
- (gst_smokedec_init), (gst_smokedec_link), (gst_smokedec_chain):
- * ext/jpeg/gstsmokedec.h:
- * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_get_type),
- (gst_smokeenc_base_init), (gst_smokeenc_class_init),
- (gst_smokeenc_init), (gst_smokeenc_getcaps), (gst_smokeenc_link),
- (gst_smokeenc_resync), (gst_smokeenc_chain),
- (gst_smokeenc_set_property), (gst_smokeenc_get_property):
- * ext/jpeg/gstsmokeenc.h:
- * ext/jpeg/smokecodec.c: (smokecodec_init_destination),
- (smokecodec_flush_destination), (smokecodec_term_destination),
- (smokecodec_init_source), (smokecodec_fill_input_buffer),
- (smokecodec_skip_input_data), (smokecodec_resync_to_restart),
- (smokecodec_term_source), (smokecodec_encode_new),
- (smokecodec_decode_new), (smokecodec_info_free),
- (smokecodec_set_quality), (smokecodec_get_quality),
- (smokecodec_set_threshold), (smokecodec_get_threshold),
- (smokecodec_set_bitrate), (smokecodec_get_bitrate),
- (find_best_size), (abs_diff), (put), (smokecodec_encode),
- (smokecodec_parse_header), (smokecodec_decode):
- * ext/jpeg/smokecodec.h:
- Added a new simple jpeg based codec
-
-2004-06-08 Wim Taymans <wim@fluendo.com>
-
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_loop):
- Fix memory leak
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_client_remove),
- (gst_tcpserversink_handle_client_read), (gst_tcp_buffer_write),
- (gst_tcpserversink_handle_client_write), (gst_tcpserversink_chain),
- (gst_tcpserversink_init_send), (gst_tcpserversink_close):
- * gst/tcp/gsttcpserversink.h:
- take streamheader into account
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/level/Makefile.am:
- * gst/level/gstlevel.c: (gst_level_class_init):
- clean up marshal generation
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
+ (gst_vorbis_enc_set_metadata):
+ Use vorbis comment utility functions from libgsttag
+ instead of re-inventing the wheel (partially fixes #347091).
- * gst/tcp/Makefile.am:
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_get_type),
- (gst_tcpclientsink_class_init), (gst_tcpclientsink_init),
- (gst_tcpclientsink_set_property), (gst_tcpclientsink_get_property):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
- (gst_tcpclientsrc_init), (gst_tcpclientsrc_set_property),
- (gst_tcpclientsrc_get_property):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_init), (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_handle_client_read),
- (gst_tcpserversink_handle_client_write),
- (gst_tcpserversink_set_property), (gst_tcpserversink_get_property):
- * gst/tcp/gsttcpserversink.h:
- add signals client-added and client-removed
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
- (gst_tcpserversrc_init), (gst_tcpserversrc_set_property),
- (gst_tcpserversrc_get_property):
- uniformized, change default protocol to NONE
- * gst/tcp/gsttcp-marshal.list: added
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- handle discont events if they happen before caps nego
-
-2004-06-07 Wim Taymans <wim@fluendo.com>
-
- * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
- (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain),
- (gst_multipart_demux_plugin_init):
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_init), (gst_multipart_mux_loop),
- (gst_multipart_mux_change_state):
- Small updates, fix a memleak
-
-2004-06-07 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * configure.ac: OSS portability
- * ext/arts/gst_arts.c: idem
- * sys/oss/gstosselement.c: idem
- * sys/oss/gstossmixer.c: idem
- * sys/oss/gstosssink.c: idem
- * sys/oss/gstosssrc.c: idem
- * sys/oss/oss_probe.c: idem
- - check for soundcard.h in different places for some BSD
-
-2004-06-07 Jan Schmidt <thaytan@mad.scientist.com>
+2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Fix leaks. Wait for state transitions that might happen ASYNC, as well
+ as some that won't.
+
+2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ Don't try to GObject scan the netbuffer as it's not a GObject.
+ Fixes #351308.
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ Document GstNetBuffer.
+
+2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (GST_START_TEST),
+ (audioconvert_suite):
+ Add testcase for caps-size-explosion
- * AUTHORS:
- Add me to the authors file
- * configure.ac:
- Increase the libdv requirement to >= version 0.100
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_src_convert),
- (gst_dvdec_src_query), (gst_dvdec_handle_sink_event),
- (gst_dvdec_push), (gst_dvdec_loop), (gst_dvdec_change_state):
- * ext/dv/gstdvdec.h:
- Add support for the new_media flag when sending DISCONT events
- Make the querying work when video pad is not linked
-
-2004-06-07 Tim-Philipp Müller <t.i.m@zen.co.uk>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init):
- create a NULL-initialized array of pads, so we don't think they
- exist already. (fixes #143130)
-
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init),
- (mixmatrix_resize), (gst_mixmatrix_set_all_caps),
- (gst_mixmatrix_request_new_pad), (gst_mixmatrix_loop):
- don't use // coments
-
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_samples_to_timestamp):
- cast to GstClockTime to get higher granularity
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- use gst_element_set_time_delay to get the exact time
- * ext/mad/gstmad.c: (gst_mad_chain):
- use the negotiated rate instead of the current frame's rate which
- might be wrong because of bit errors. This avoids emitting totally
- bogus timestamps and screwing sync.
- (fixes #143454)
-
-2004-06-07 Tim-Philipp Müller <t.i.m@zen.co.uk>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * gst/adder/gstadder.c: (gst_adder_loop):
- properly error out when no negotiation has happened yet. (fixes
- #143032)
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- forward correctly transformed offset in discont events. Based on
- patch by Arwed v. Merkatz. (fixes #142851)
-
-2004-06-06 David Schleef <ds@schleef.org>
-
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c: that's
- G_HAVE_GNUC_VARARGS, not G_HAVE_GNU_VARARGS. Should fix compile
- problems on several systems.
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- use explicit caps on the srcpad
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- properly error out if caps couldn't be set (fixes #142764)
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
- (gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
- (gst_alsa_start_audio):
- - don't call set_periods_integer anymore, it breaks the
- configuration randomly
- - call snd_pcm_hw_params_set_access directly instead of using masks
- - don't fail if the sw_params can't be set, just use the default
- params and hope it works. Alsalib has weird issues when you touch
- sw_params and does no proper error reporting about what failed.
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio),
- (gst_alsa_close_audio):
- make our alsa debugging go via gst debugging and not conditionally
- defined
- * ext/alsa/gstalsa.h:
- add ALSA_DEBUG_FLUSH macro
- * ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper),
- (plugin_init):
- wrap alsa errors to be printed via the gst debugging system and not
- spammed to stderr
-
-2004-06-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
- (gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
- (gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (qtdemux_dump_mvhd),
- (qtdemux_parse_trak):
- * gst/qtdemux/qtdemux.h:
- Bitch. Also known as seeking, querying & co.
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- * sys/oss/gstosssink.h:
- Resyncing is for weenies, this hack is no longer needed and was
- broken anyway (since it - unintendedly - always leaves resync to
- TRUE).
-
-2004-06-05 Andrew Turner <zxombie@hotpop.com>
-
- * gst/tcp/gsttcp.c: portability (Solaris 10/FreeBSD)
- * gst/tcp/gsttcpclientsrc.h: idem
- - define MSG_NOSIGNAL if not done
- - include unistd.h for off_t
- (fixes #143749)
-
-2004-06-05 Benjamin Otte <otte@gnome.org>
+2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- * ext/kio/Makefile.am:
- check for qt's moc preprocessor explicitly and use it
+ gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_get_unit_size), (set_structure_widths):
+ Lower debug, use g_assert in _get_unit_size
+ * gst/audioresample/gstaudioresample.c:
+ (audioresample_get_unit_size):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_get_unit_size):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
+ use g_assert in _get_unit_size
-2004-06-03 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
- don't get a signal for EPIPE on socket writes
- (somebody check if this works on other platforms)
+ * ChangeLog:
+ ChangeLog surgery: fix bug number
+ Original commit message from CVS:
+ ChangeLog surgery: fix bug number
+
+2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Document GstRTPBuffer.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
+ (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
+ (gst_rtp_buffer_get_payload_buffer):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Document GstRTPBuffer.
+ Added function to efficiently strip payload headers.
+ API: gst_rtp_buffer_get_payload_subbuffer()
+
+2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
+ (gst_tag_to_vorbis_comments):
+ Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
+ tags and deserialise them properly as well (#351768).
+ Add some more gtk-doc blurbs and also some g_return_if_fail().
+ * tests/check/libs/tag.c: (GST_START_TEST),
+ (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
+ More tests.
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
+ Original commit message from CVS:
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstogg.c: (plugin_init):
+ * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
+ (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
+ (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
+ (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
+ (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
+ (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
+ Added ogg-in-avi parser element. Fixes #140139.
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
+ Fixed a bug in oggdemux debug code.
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Recognise Ogg in the AVI extensible wave format.
+
+2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ Make buffer durations add up (duration should be next_ts-ts for
+ perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
+ from CVS.
+ * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
+ (test_buffer_timestamps), (cddabasesrc_suite):
+ Add unit test for the above.
+ * tests/check/Makefile.am:
+ Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
+ to see what happens.
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_get_type):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- check error condition on available samples correctly
+2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/alsa/: Avoid setting and using a NULL device name.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
+ (gst_alsasink_open):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
+ (gst_alsasrc_open):
+ Avoid setting and using a NULL device name.
+ Print more info when we fail to open a device.
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_get_time):
- avoid a segfault
- * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
- (gst_tcp_gdp_read_header), (gst_tcp_gdp_read_caps):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_gdp_read_caps),
- (gst_tcpserversrc_gdp_read_header):
- use ssize_t over size_t since the former is signed and thus the
- check for error codes can work
+2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-02 Wim Taymans <wim@fluendo.com>
+ API: add gst_tag_parse_extended_comment() (#351426).
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
+ API: add gst_tag_parse_extended_comment() (#351426).
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
+ Add unit test for gst_tag_parse_extended_comment().
- reviewed by: Johan
+2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_loop):
- Oops
+ sys/: Fix leak (#351502).
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
+ Fix leak (#351502).
-2004-06-02 Wim Taymans <wim@fluendo.com>
+2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_init), (gst_multipart_mux_loop),
- (gst_multipart_mux_get_property), (gst_multipart_mux_set_property),
- (gst_multipart_mux_change_state):
- Added configurable boundary specifier, added the value as a
- caps field as well.
+ Document playbin.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * gst/playback/gstplaybin.c:
+ Document playbin.
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Update to CVS version.
-2004-06-02 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversrc.c:
- - portability fix, to compile on OSX
- (fixes #143146)
+ gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_property), (gst_play_bin_get_property),
+ (value_list_append_structure_list),
+ (gst_play_bin_handle_redirect_message),
+ (gst_play_bin_handle_message):
+ Add "connection-speed" property; re-order redirect messages with
+ multiple redirect locations depending on the minimum bitrate if
+ that information is available and a connection speed is set
+ (#350399).
- * sys/osxaudio/gstosxaudioelement.c:
- * sys/osxaudio/gstosxaudiosink.c:
- * sys/osxaudio/gstosxaudiosrc.c:
- - compilation warnings on OSX
- (fixes #143153)
+2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-02 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ Update max volume to the same value that the volume element uses.
- * ext/vorbis/vorbisdec.c : sign warning fixes
+2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsamixer.c: Less uglyness..
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
+ Less uglyness..
+
+2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Add some more debug info.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
+ (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
+ Add some more debug info.
+ Don't crash when a seek failed.
+ Actually return the result of the seek instead of TRUE.
+ Ignore multiple BOS pages with the same serial so that we don't create
+ the same stream multiple times.
+ Post an error when we fail to do the initial seek.
+
+2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsa.c: Small code cleanup.
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
+ (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
+ Small code cleanup.
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
+ (gst_alsa_mixer_new):
+ Remove hack that always set the device to hw:0*.
+ Properly find the card name for whatever device was configured.
+ Do some better debugging.
+ Fixes #350784.
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_set_property),
+ (gst_alsa_mixer_element_change_state):
+ Cleanups.
+ Handle setting of a NULL device name better.
+
+2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Don't clip float values. Fixes #350900.
+
+2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/tcp/gsttcp.c: Really fix the build?
+ Original commit message from CVS:
+ 2006-08-11 Andy Wingo <wingo@pobox.com>
+ * gst/tcp/gsttcp.c: Really fix the build?
+
+2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
+ Original commit message from CVS:
+ 2006-08-11 Andy Wingo <wingo@pobox.com>
+ * gst/tcp/gsttcp.h: For now, always disable deprecation here --
+ fixes the build.
+
+2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
+ Float caps shouldn't have a "signed" field.
- * gst-libs/gst/mixer/mixertrack.c :
- do no use defines which are glib 2.4 specific
+2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-01 Christophe Fergeau <teuf@gnome.org>
+ ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
+ Implement SEEKING query in its most basic form, so that we can
+ at least check if we're seekable or not (#350655).
- * ext/flac/gstflactag.c: strip ending framing bit from vorbiscomment
- buffer since libflac doesn't expect it (reports a sync error when
- it encounters that)
+2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
+ gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
+ The checks here are not even close to anything that would
+ justify MAXIMUM probability, lowering to POSSIBLE until someone
+ fixes the checks (case at hand: quicktime redirection files
+ might start with 00 00 01 XX and pass the checks here just
+ fine, see #350399).
-2004-06-01 Owen Fraser-Green <owen@discobabe.net>
+2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/mixer/mixertrack.h: Changed struct syntax
- * gst-libs/gst/mixer/mixertrack.c:
- (gst_mixer_track_get_property), (get_mixer_track_init),
- (get_mixer_track_get_property): Added property accessors
- * gst-libs/gst/mixer/mixeroptions.h: Changed struct syntax
- * gst-libs/gst/mixer/mixeroptions.c:
- (gst_mixer_options_get_values): Added
- * gst-libs/gst/mixer/mixer.h: Changed GstMixerClass syntax
- * gst-libs/gst/mixer/mixer.c: Fixed comment
+ tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
+ I forgot to include the file containing the #define :)
+ Now includes "config.h"
+2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-06-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
+ Ignore test known to fail on PPC64. See #348114.
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio):
- improve error messages on open
+2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
+ gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
+ Better detection for multipart/x-mixed-replace: accept leading
+ whitespaces before the boundary marker as well (as our very own
+ multipartmux used to produce) (#349068).
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- check if v4l-conf is in path
+ gst-libs/gst/riff/: Detect DTS audio streams (#350157).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Detect DTS audio streams (#350157).
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
+ Original commit message from CVS:
+ 2006-08-05 Andy Wingo <wingo@pobox.com>
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c (gst_theora_parse_class_init)
+ (theora_parse_dispose, theora_parse_set_property)
+ (theora_parse_get_property, theora_parse_munge_granulepos)
+ (theora_parse_push_buffer, theora_parse_change_state): Add a
+ property 'synchronization-points' to fix badly synchronized oggs.
+
+2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
+
+ gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
+ Original commit message from CVS:
+ 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ Fix event parsing by gdpdepay. Fixes #349916.
+
+2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/: Add a few tests for the channel position stuff in libgstaudio.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/audio.c: (structure_contains_channel_positions),
+ (fixed_caps_have_channel_positions), (GST_START_TEST),
+ (audio_suite), (main):
+ Add a few tests for the channel position stuff in libgstaudio.
- * gst-libs/gst/media-info/media-info-priv.c: (gmi_set_mime):
- change assert to a more readable error message
+2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-31 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
+ (gst_alsa_detect_channels):
+ * ext/alsa/gstalsasink.c:
+ Add support for cards that (only) do more than 8 channels,
+ like the Delta 44 (#345188).
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions):
+ * gst-libs/gst/audio/multichannel.h:
+ API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
+ unspecified channel position and cannot be combined with any
+ of the other audio channel positions; adjust position layout
+ checks accordingly (#345188).
- * gst-libs/gst/tuner/tunerchannel.h:
- - add a freq_multiplicator field to make the conversion
- between internal frequency unit and Hz
- * sys/v4l/gstv4lelement.c:
- * sys/v4l2/gstv4l2element.c:
- - change default video device to /dev/video0
- * sys/v4l/v4l_calls.c:
- * sys/v4l2/v4l2_calls.c:
- - we only expose frequency to the user in Hz instead of
- bastard v4lX unit (either 62.5kHz or 62.5Hz)
+2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-31 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Initialise b_o_s and e_o_s variables
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add some unusual fourcc's from mplayer avi's
- * gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
- Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
- autoplugging.
+ gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Recognise ancient RealAudio files (see #349779).
-2004-05-28 Wim Taymans <wim@fluendo.com>
+2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
- * configure.ac:
- * gst/alpha/Makefile.am:
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_get_type), (gst_alpha_base_init),
- (gst_alpha_class_init), (gst_alpha_init), (gst_alpha_set_property),
- (gst_alpha_get_property), (gst_alpha_sink_link), (gst_alpha_add),
- (gst_alpha_chroma_key), (gst_alpha_chain),
- (gst_alpha_change_state), (plugin_init):
- A plugin to add an alpha channel to I420 video. Can optionally do
- chroma keying.
- * gst/multipart/Makefile.am:
- * gst/multipart/multipart.c: (plugin_init):
- * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
- (gst_multipart_demux_class_init), (gst_multipart_demux_init),
- (gst_multipart_demux_finalize), (gst_multipart_demux_handle_event),
- (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain),
- (gst_multipart_demux_change_state),
- (gst_multipart_demux_plugin_init):
- * gst/multipart/multipartmux.c: (gst_multipart_mux_get_type),
- (gst_multipart_mux_base_init), (gst_multipart_mux_class_init),
- (gst_multipart_mux_get_sink_event_masks), (gst_multipart_mux_init),
- (gst_multipart_mux_sinkconnect), (gst_multipart_mux_pad_link),
- (gst_multipart_mux_pad_unlink),
- (gst_multipart_mux_request_new_pad),
- (gst_multipart_mux_handle_src_event),
- (gst_multipart_mux_next_buffer), (gst_multipart_mux_compare_pads),
- (gst_multipart_mux_queue_pads), (gst_multipart_mux_loop),
- (gst_multipart_mux_get_property), (gst_multipart_mux_set_property),
- (gst_multipart_mux_change_state), (gst_multipart_mux_plugin_init):
- A Multipart demuxer/muxer. Not sure if it violates specs. Used to
- send multipart jpeg images to a browser.
- * gst/videobox/Makefile.am:
- * gst/videobox/README:
- * gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type),
- (gst_video_box_get_type), (gst_video_box_base_init),
- (gst_video_box_class_init), (gst_video_box_init),
- (gst_video_box_set_property), (gst_video_box_get_property),
- (gst_video_box_sink_link), (gst_video_box_i420),
- (gst_video_box_ayuv), (gst_video_box_chain),
- (gst_video_box_change_state), (plugin_init):
- Crops or adds borders around an image. can do alpha channel
- borders as well.
- * gst/videomixer/Makefile.am:
- * gst/videomixer/README:
- * gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
- (gst_videomixer_pad_base_init), (gst_videomixer_pad_class_init),
- (gst_videomixer_pad_get_sink_event_masks),
- (gst_videomixer_pad_get_property),
- (gst_videomixer_pad_set_property),
- (gst_videomixer_pad_sinkconnect), (gst_videomixer_pad_link),
- (gst_videomixer_pad_unlink), (gst_videomixer_pad_init),
- (gst_video_mixer_background_get_type), (gst_videomixer_get_type),
- (gst_videomixer_base_init), (gst_videomixer_class_init),
- (gst_videomixer_init), (gst_videomixer_request_new_pad),
- (gst_videomixer_handle_src_event),
- (gst_videomixer_blend_ayuv_i420), (gst_videomixer_fill_checker),
- (gst_videomixer_fill_color), (gst_videomixer_fill_queues),
- (gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
- (gst_videomixer_loop), (gst_videomixer_get_property),
- (gst_videomixer_set_property), (gst_videomixer_change_state),
- (plugin_init):
- Generic video mixer plugin, can handle multiple inputs all with
- different framerates and video sizes. Is fully alpha channel
- aware.
-
-2004-05-27 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- Select first track as master track. Not sure how else to handle
- that...
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer):
- Discard discont events. Should fix #142962.
-
-2004-05-26 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
+ Original commit message from CVS:
+ Patch by: Jens Granseuer <jensgr at gmx net>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinder for Interplay's MVE format (#348973).
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init),
- (gst_alsa_mixer_build_list), (gst_alsa_mixer_get_volume),
- (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
- (gst_alsa_mixer_set_record), (gst_alsa_mixer_set_option),
- (gst_alsa_mixer_get_option):
- * ext/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixeroptions.c:
- (gst_alsa_mixer_options_get_type),
- (gst_alsa_mixer_options_class_init), (gst_alsa_mixer_options_init),
- (gst_alsa_mixer_options_new):
- * ext/alsa/gstalsamixeroptions.h:
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- * ext/alsa/gstalsamixertrack.h:
- Add enumerations (as GstMixerOptions). Make correct distinction
- between input/output tracks. Add capture/playback private flag.
- Use flag to decide on whether to set capture or playback volumes
- or switches. Use playback and record switches.
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/mixer/mixer-marshal.list:
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init),
- (gst_mixer_set_option), (gst_mixer_get_option),
- (gst_mixer_mute_toggled), (gst_mixer_record_toggled),
- (gst_mixer_volume_changed), (gst_mixer_option_changed):
- * gst-libs/gst/mixer/mixer.h:
- * gst-libs/gst/mixer/mixeroptions.c: (gst_mixer_options_get_type),
- (gst_mixer_options_class_init), (gst_mixer_options_init),
- (gst_mixer_options_dispose):
- * gst-libs/gst/mixer/mixeroptions.h:
- Add GstMixerOptions.
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Rename Audio Mixer to OSS Mixer (similar to Alsa Mixer). Fix
- broken device detection on computers with multiple OSS sound
- cards.
-
-2004-05-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
- fixate nicely even when the peer is not negotiating
-
-2004-05-25 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
+ Original commit message from CVS:
+ Patch by: Marcel Moreaux <marcelm at luon dot net>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_add_to_queue):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Handle RTP sequence number rollover.
+ Disable jitterbuffer by default.
+
+2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/gdp/gstgdpdepay.c: Disable seeking.
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
+ (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
+ (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
+ (gst_gdp_depay_change_state):
+ Disable seeking.
+ Small cleanups.
+ Clear adapter on disconts.
+ Clear caps when going to READY instead of NULL
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
+ (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
+ (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
+ (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
+ (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
+ (gst_gdp_pay_change_state):
+ * gst/gdp/gstgdppay.h:
+ Reset payloader when going to READY.
+ Fix leaked buffers in ->queue on push errors.
+ Disable seeking.
+ Code cleanups.
+ Create packetizer in _init, free in _finalize.
+
+2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
+ (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
+ Consume all events except EOS because we generate events from
+ the gdp payload instead. Fixes #349204
+
+2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (audioresample_stop),
+ (audioresample_set_caps):
+ Don't leak references to the incoming caps. Clean them up when
+ stopping.
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
+ (gst_video_scale_finalize):
+ Don't leak our temporary pixel buffer.
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
+ (GST_START_TEST), (simple_launch_lines_suite):
+ Fix leaks and re-enable the test for valgrind checking.
+
+2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
+ (plugin_init):
+ Add typefind function for multipart/x-mixed-replace (#348916).
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps):
- make sure we don't allow depth > width
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
- fixate endianness to G_BYTE_ORDER as default
- * gst/audioscale/gstaudioscale.c:
- we don't handle another endianness as host-endianness
+2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-25 David Schleef <ds@schleef.org>
+ gst/adder/gstadder.c: Fix leak in duration query.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps),
+ (gst_adder_query_duration):
+ Fix leak in duration query.
+ Reflow some docs and notes.
- * gst/ffmpegcolorspace/mem.c: malloc() is in stdlib.h, not malloc.h
+2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
-2004-05-24 Benjamin Otte <otte@gnome.org>
+ tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
+ Original commit message from CVS:
+ * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
+ (vorbisenc_suite):
+ Enable Andy's extra vorbisenc test, now that it passes. Also fix one
+ aspect of it.
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
- (gst_oggvorbisenc_setup):
- properly fail when we can't setup the vorbis encoder due to
- unsupported settings
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
- (gst_vorbisenc_setup):
- same
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- fix case where warnings occured when one pad was unlinked while the
- other's link function was called
+2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
-2004-05-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
+ (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
+ (gst_vorbis_enc_push_buffer),
+ (gst_vorbis_enc_buffer_check_discontinuous),
+ (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
+ * ext/vorbis/vorbisenc.h:
+ Handle discontinuities in the input vorbis stream correctly,
+ so that the output is properly timestamped (and has good granulepos
+ values). Needs some oggmux fixes too.
- * gst/tcp/Makefile.am:
- use GST_ENABLE_NEW
+2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
+ Original commit message from CVS:
+ patch by: Kai Vehmanen <kv2004 eca cx>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_handle_sink_event),
+ (gst_base_rtp_depayload_change_state):
+ Don't send multiple newsegments with different formats.
+ Fixes #348677.
+
+2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
+ (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
+ Make seeking in ogg more accurate again by doing the more correct
+ granuletime to stream time conversion.
+
+2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_new_client):
+ debug a little more understandably
+ do not use goto as a substitute for break, especially if
+ break is also being used
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/resample/private.h:
- don't use optimizations that are #if 0'ed
+ * gst/tcp/gsttcp.c:
+ move a recurring normal event to LOG, where it should be
+ Original commit message from CVS:
+ move a recurring normal event to LOG, where it should be
-2004-05-24 Wim Taymans <wim@fluendo.com>
+2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
- Fix potential division by zero error and hopefully get
- the position query right to get correct timestamps on avi
- audio.
+ * ext/vorbis/vorbisdec.c:
+ tweak debug output
+ Original commit message from CVS:
+ tweak debug output
+
+2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
+ proxying get/set caps is the wrong thing to do, since we really
+ do change caps quite fundamentally
+ * tests/check/elements/gdpdepay.c:
+ * tests/check/elements/gdppay.c:
+ remove declaration of buffers, it's already done in gstcheck.h
+
+2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/: Remove GLib-2.6 compatibility cruft.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
+ Remove GLib-2.6 compatibility cruft.
-2004-05-24 Wim Taymans <wim@fluendo.com>
+2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Don't try to align a sample to an unknown value.
+
+2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
+ When the audio clock is slaved to another clock, never try to align
+ samples but trust the rate interpolation algorithm.
+
+2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
+ Don't try to calculate silence samples, base class does this much
+ better now.
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
+ (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
+ (gst_ring_buffer_acquire):
+ Calculate silence samples correctly.
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Add _CAST macro.
+
+2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
+ Limit search for the first markup tag to the first few kB of
+ the file. If we don't find one there, it's highly unlikely that
+ this is an XML(-ish) file.
- * gst/videoscale/videoscale.c: (gst_videoscale_scale_nearest),
- (gst_videoscale_scale_nearest_str2),
- (gst_videoscale_scale_nearest_str4),
- (gst_videoscale_scale_nearest_32bit),
- (gst_videoscale_scale_nearest_24bit),
- (gst_videoscale_scale_nearest_16bit):
- Fix the scaling algorithm and avoid a buffer overflow.
- removed the while loop in the scaling function as it
- was used for point sampling only.
+2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
+ Original commit message from CVS:
+ 2006-07-21 Andy Wingo <wingo@pobox.com>
+ * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
+ test to the one in vorbisenc. Also commented out.
+
+2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
+ Original commit message from CVS:
+ 2006-07-21 Andy Wingo <wingo@pobox.com>
+ * tests/check/pipelines/vorbisenc.c:
+ (test_discontinuity): New test, commented out until Mike lands
+ some elite vorbisenc patches.
+
+2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/pipelines/: Port to bufferstraw.
+ Original commit message from CVS:
+ 2006-07-21 Andy Wingo <wingo@pobox.com>
+ * tests/check/pipelines/vorbisenc.c:
+ * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
+ Bufferstraw was actually factored out of these tests. Now we share
+ code yay.
+
+2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Better clipping.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (clip_buffer):
+ Better clipping.
+
+2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Fix leak.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
+ (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
+ (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
+ Fix leak.
+ Avoid type casting when we can.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
+ Fix mem leak.
+
+2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_change_state):
+ Make state change fail if the specified device can't be opened
+ for some reason.
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/mad/gstid3tag.c: (gst_id3_tag_get_type),
- (gst_id3_tag_class_init), (gst_id3_tag_init),
- (gst_id3_tag_set_property), (gst_id3_tag_get_tag_to_render),
- (gst_id3_tag_handle_event), (gst_id3_tag_do_caps_nego),
- (gst_id3_tag_send_tag_event):
- lots of fixes to make id3mux work and id3demux work correctly
+ gst/playback/test.c: Example of a small audio/video player using decodebin.
+ Original commit message from CVS:
+ * gst/playback/test.c: (gen_video_element), (gen_audio_element),
+ (cb_newpad), (main):
+ Example of a small audio/video player using decodebin.
-2004-05-24 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/Makefile.am:
- add rules to build shout2send (was removed by accident
- when this module was no more marked experimental/broken)
+ gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add 'fact' chunk id
-2004-05-24 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_change_state):
+ Don't assert when not negotiated but post a meaningfull
+ error message. Fixes #347918.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ Add comment about better default MTU size.
+ * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
+ Small cleanups, start docs.
+
+2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
+
+ sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
+ Original commit message from CVS:
+ Patch by: Martin Szulecki
+ * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
+ If "device-name" is requested and the device is not
+ open, try to temporarily open it to obtain this
+ information (#342494).
- * ext/shout2/gstshout2.c:
- * ext/shout2/gstshout2.h:
- adding a "connection problem" signal to shout2send
- (fixes #142954)
+2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Some more random const-ifications.
- * ext/kio/kioreceiver.cpp:
- * ext/kio/kioreceiver.h:
- fix sign comparison issues
+2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ (gst_riff_create_video_template_caps):
+ Add more FOURCCs (sort list to make stuff easier to find),
+ add comment what those 16 bytes in struct _gst_riff_strh according to
+ one avi-dumper are
- * gst/cdxaparse/gstcdxaparse.c:
- * gst/cdxaparse/gstcdxaparse.h:
- some renaming
- add some checks/sanity
- prepare for seek addition
+2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/sunaudio/gstsunaudio.c:
- remove exported dupe init function
+ gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
+ Original commit message from CVS:
+ 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
+ remove parent_class setting, BOILERPLATE does this
+ (gst_gdp_pay_reset_streamheader):
+ fix typo in comment
-2004-05-21 Jan Schmidt <thaytan@mad.scientist.com>
+2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_get_formats),
- (gst_dvdec_src_convert), (gst_dvdec_sink_convert):
- Fix format conversion and position querying.
- * gst/debug/progressreport.c: (gst_progressreport_report):
- Don't output a bogus total value that we didn't query.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Always set XV_AUTOPAINT_COLORKEY to true. Fixes xvimagesink showing
- only a blank window after xine has been used.
+ gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions),
+ (gst_audio_fixate_channel_positions):
+ Const-ify two arrays.
-2004-05-21 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
- * m4/as-arts.m4:
- sync with upstream version to fix test on FC2
- readd with -ko to preserve Id header
+ ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
+ Fix typo, so that alsasink also advertises 8 channels
+ if that's supported (tags: can, worms, open, alsa, ph34r).
-2004-05-20 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
+ (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
+ *sigh*, when is the compiler going to warn when the comments
+ are out-of-sync with the code.. Refix case of busted theora
+ headers with 0 granule pos.
+
+2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_wait),
+ (gst_base_rtp_depayload_change_state),
+ (gst_base_rtp_depayload_set_property),
+ (gst_base_rtp_depayload_get_property):
+ Fix 99% cpu load by waiting for absolute times on the
+ clock. Fixes #347300.
+
+2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
+ Original commit message from CVS:
+ 2006-07-14 Andy Wingo <wingo@pobox.com>
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
+ (theora_parse_push_headers, theora_parse_clear_queue)
+ (theora_parse_drain_queue_prematurely, )
+ (theora_parse_sink_event, theora_parse_change_state): Queue events
+ until we initialized our state, like in vorbisparse.
+
+2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
+
+ ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
+ Original commit message from CVS:
+ 2006-07-14 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.h:
+ * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
+ (vorbis_parse_push_headers, vorbis_parse_clear_queue)
+ (vorbis_parse_drain_queue_prematurely, )
+ (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
+ until we have initialized our state. Fixes seeking after an
+ initial pad block.
+ 2006-07-14 Andy Wingo <wingo@pobox.com>
+ Patch by: Iain * <iaingnome@gmail.com>
+ * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
+
+2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Bump nano back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Bump nano back to CVS
- * configure.ac:
- test for FIONREAD ioctl in sys/filio.h for Solaris compat.
- * gst/tcp/gsttcpclientsrc.c: idem
- * gst/tcp/gsttcpserversink.c: idem
- * gst/tcp/gsttcpserversrc.c: idem
- * m4/gst-fionread.m4: idem
+=== release 0.10.9 ===
- * sys/sunaudio/gstsunaudio.c: change category to Sink/Audio
+2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac: enable speex plugin for speex 1.1.5+
- * ext/speex/gstspeexenc.c: fix cast warning
+ configure.ac: releasing 0.10.9, "I walk the line"
+ Original commit message from CVS:
+ 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ releasing 0.10.9, "I walk the line"
- * ext/esd/README: fix typo
+2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
-2004-05-20 David Schleef <ds@schleef.org>
+ tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
+ Original commit message from CVS:
+ * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
+ Move a g_cond_signal to earlier to avoid sometimes deadlocking
+ (commonly happens when running this test under valgrind) when trying
+ to remove the buffer probe.
- * configure.ac: Minor cosmetic change to convince the buildbot to
- reautogen.
- * sys/sunaudio/gstsunaudio.c: (gst_sunaudiosink_class_init),
- (gst_sunaudiosink_init), (gst_sunaudiosink_getcaps),
- (gst_sunaudiosink_pad_link), (gst_sunaudiosink_chain),
- (gst_sunaudiosink_setparams), (gst_sunaudiosink_open),
- (gst_sunaudiosink_close), (gst_sunaudiosink_change_state),
- (gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property):
- More hacking. Plays audio now.
+2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-20 David Schleef <ds@schleef.org>
+ * gst/gdp/Makefile.am:
+ build as a plugin, not a lib
+ Original commit message from CVS:
+ build as a plugin, not a lib
+
+2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
+ Fix missing g_unlock from the previous commit
+
+2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
+ (gst_ximagesink_change_state):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_change_state):
+ Implement a locking order to ensure we always take the object lock
+ before the x_lock and never vice-versa.
+
+2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ docs/plugins/: add more plugins and elements to docs
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ add more plugins and elements to docs
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ fix segfaults due to wrong g_free
+ add example
+ * gst/gdp/gstgdppay.c:
+ add example
+
+2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (find_compatibles):
+ Fix a caps leak when linking (#347304)
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
+ (gst_ximagesink_change_state):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
+ (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
+ Don't leak shared memory resources. Use the object lock to protect
+ against the xcontext disappearing while returning a buffer from the
+ pipeline. (#347304)
+
+2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
+ (vorbis_handle_comment_packet):
+ gst_tag_list_merge() returns a new object. Take that into account when
+ using it. This avoids memleak.
+ Revert previous commit which is not needed.
+
+2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
+ Reset the decoder in finalize so that all fields get cleared.
+
+2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_set_clock),
+ (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
+ Don't try to post an error message when setting the clock fails
+ as this can happen when adding an element to a bin which will then
+ deadlock. Fixes #347296.
+
+2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
+ (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
+ (vorbis_handle_type_packet):
+ Post tag messages on the bus even if we're not initialized.
+ If we're not initialized, we still postpone the event pushing of tags.
+
+2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Revert last two changes that broke the freeze.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
+ (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
+ Revert last two changes that broke the freeze.
+
+2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
+ basesink calculates silence sample correctly for us.
+
+2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
+ (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
+ Calculate correct silence samples so we don't fill our ringbuffer
+ with noise.
+
+2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
+ (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
+ (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
+ * ext/vorbis/vorbisdec.h:
+ Delay sending events (newsegment, tags) until the decoder is properly
+ initialized.
+ Fixes #347295
+
+2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- * sys/Makefile.am:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-05-20 David Schleef <ds@schleef.org>
+2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/osxaudio/Makefile.am: New OS X audio plugin by Zaheer Abbas Merali
- * sys/osxaudio/gstosxaudio.c:
- * sys/osxaudio/gstosxaudioelement.c:
- * sys/osxaudio/gstosxaudioelement.h:
- * sys/osxaudio/gstosxaudiosink.c:
- * sys/osxaudio/gstosxaudiosink.h:
- * sys/osxaudio/gstosxaudiosrc.c:
- * sys/osxaudio/gstosxaudiosrc.h:
+ tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (get_float_mc_caps),
+ (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
+ Patch from #347221 adding a test for audioconvert
+ channel remappings.
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps),
- (gst_vorbisenc_chain):
- put the codec headers on the caps as streamheader as well as
- pushing them out
+ gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
+ Original commit message from CVS:
+ * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
+ (gst_ssa_parse_parse_line):
+ Don't include the terminating NUL in the buffer size,
+ it's only there for extra paranoia (would add random
+ '*' characters at the end of each subtitle since the
+ terminator itself is not valid UTF-8 technically).
+ Also fix indenting after boilerplate macro.
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisenc.c: (vorbis_granule_time_copy),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
- (gst_vorbisenc_push_packet), (gst_vorbisenc_chain):
- split up push_packet into two functions
+ gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ Also emit 'unknown-type' signal (which should really be
+ called unhandled-type) if we found potential decoders/demuxers
+ in the registry but none of them worked in the end (as in the
+ case where the plugins don't exist any longer but are still
+ listed in the registry). Fixes #329798.
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
- * gst/tcp/.cvsignore:
- ignore enums
- * gst/tcp/Makefile.am:
- * gst/tcp/README:
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpplugin.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c:
- * gst/tcp/gsttcpserversrc.h:
- add new tcp elements
+ * ChangeLog:
+ * ext/theora/theoraparse.c:
+ theoraparse.c (theora_parse_push_buffer)
+ Original commit message from CVS:
+ 2006-07-08 Andy Wingo <wingo@pobox.com>
+ * theoraparse.c (theora_parse_push_buffer)
+ (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
+ Add some more debugging. Fix granulepos reconstruction in the face
+ of discontinuities.
+
+2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init),
+ (gst_base_audio_sink_provide_clock):
+ Use gobject_class instead of G_OBJECT_CLASS (klass)
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init), (gst_base_audio_src_init),
+ (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
+ (gst_base_audio_src_get_time),
+ (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
+ (gst_base_audio_src_create_ringbuffer):
+ Fix latency and buffer-time constants and properties ala basesink.
+ Implement pull based scheduling. Fixes #346527.
+ Set default blocksize in GstBaseSrc to 0, we default to pushing out
+ one segment.
+ Refuse slaving to another clock instead of silently not working.
+ Only provide a clock when we are actually able to do so.
+ Various small cleanups and compiler hints.
+
+2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
+ Original commit message from CVS:
+ Patch by: Lutz Mueller <lutz at topfrose de>
+ * gst/typefind/gsttypefindfunctions.c: (html_type_find),
+ (plugin_init):
+ Add typefinding for text/html (#346581).
-2004-05-19 Wim Taymans <wim@fluendo.com>
+2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/law/mulaw-conversion.c: (mulaw_encode):
- Fix overflow bug in ulaw encoding.
+ gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
+ (xml_check_first_element), (xml_type_find), (smil_type_find):
+ Fix SMIL typefinding, make xml_check_first_element() more
+ useful.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- don't unref the event twice
+ gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
+ (gst_play_base_bin_finalize), (decodebin_element_added_cb),
+ (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
+ * gst/playback/gstplaybasebin.h:
+ Protect list of elements with a subtitle-encoding property and
+ the subtitle encoding member itself with a lock of their own
+ instead of using the object lock. This prevents a dead-lock in
+ the element-remove callback in some circumstances when shutting
+ down playbin.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
- * configure.ac:
- remove -Wno-sign-compare
+ win32/common/libgsttag.def: Export some new functions.
+ Original commit message from CVS:
+ * win32/common/libgsttag.def:
+ Export some new functions.
+ * win32/vs6/libgstogg.dsp:
+ Add a link to libgsttag-0.10.lib.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- remove -DG_DISABLE_DEPRECATED. It's not usable without workarounds
- if you want to work against glib 2.2 and 2.4
+ ext/alsa/gstalsamixertrack.c: Some const-ification.
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
+ Some const-ification.
-2004-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tcp/Makefile.am:
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpsink.h:
- * gst/tcp/gsttcpsrc.h:
- gsttcp -> gsttcpplugin + CVS surgery in preparation for tcp merge
+ gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
+ Improve checking if we are dealing with a stream. Added some
+ more uris that need buffering.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/debug/tests.c: (md5_get_value):
- fix segfault on gst-inspect
+ ext/vorbis/vorbisdec.c: Remove unused variable.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
+ Remove unused variable.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/debug/testplugin.c:
- * gst/debug/tests.c:
- * gst/debug/tests.h:
- add new extensible and configurable testing element. Current tests
- include buffer count, stream length, timestamp/duration matching and
- md5.
- * gst/debug/Makefile.am:
- * gst/debug/gstdebug.c: (plugin_init):
- add infrastructure for new element
+ Makefile.am: include lcov.mak
+ Original commit message from CVS:
+ * Makefile.am:
+ include lcov.mak
+ * configure.ac:
+ add GCOV_LIBS to GST_LIBS
-2004-05-19 Johan Dahlin <johan@gnome.org>
+2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
- * ext/dv/gstdvdec.c (gst_dvdec_quality_get_type): Add proper
- ending of the array. Fixes gst-inspect segfault on ppc.
+ ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
+ Original commit message from CVS:
+ Patch by: Michael Sheldon <webmaster at mikeasoft com>
+ * ext/alsa/gstalsasrc.c:
+ Add 32 bps to template caps and increase channels range
+ from [1,2] to [1,MAX]. See #346326.
-2004-05-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dirac/gstdiracdec.cc : change category to Codec/Decoder/Video
+ gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Recognise 'WMVA' video codec fourcc (#345879).
- * m4/a52.m4 : don't fix a test that should fail with current a52dec lib
+2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-05-18 David Schleef <ds@schleef.org>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ Fixed nasty memory leak
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert): Fixes for
- warnings (bugs, actually) noticed by gcc but not forte.
+2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-18 David Schleef <ds@schleef.org>
+ gst/tcp/gsttcp.c: fix logging
+ Original commit message from CVS:
+ * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
+ (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
+ fix logging
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: New sunaudiosink
+2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-05-18 David Schleef <ds@schleef.org>
+ gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
+ (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
+ Protect remove_fakesink using a mutex, so that we don't try and
+ remove the fakesink simultaneously from multiple threads.
+ When going from READY to PAUSED, restore the fakesink, so that
+ it is there when decodebin gets reused.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header): Patch from dcm@acm.org (David Moore)
- to allow qtdemux to use non-seekable streams. (bug #142272)
+2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 David Schleef <ds@schleef.org>
+ Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * sys/v4l/gstv4ljpegsrc.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lsrc.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
- * gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16),
- (gst_resample_sinc_ft_float): Remove use of static temporary
- buffer. This code was obviously not supposed to last long, but
- it's stuck in our ABI, so it required a little hack to make it
- ABI-compatible. Fixes #142585.
- * gst-libs/gst/resample/resample.h: same.
+2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
+ Original commit message from CVS:
+ * ext/directfb/dfbvideosink.c:
+ * ext/gsm/gstgsmdec.c:
+ * ext/gsm/gstgsmenc.c:
+ * ext/libmms/gstmms.c:
+ * ext/neon/gstneonhttpsrc.c:
+ * ext/theora/theoradec.c:
+ * gst/freeze/gstfreeze.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * sys/glsink/glimagesink.c:
+ Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
+ and fix one GObject boilerplate macro.
+
+2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
+ Second field in GEnumValue shouldn't be a description,
+ but a stringified version of the enum value.
-2004-05-18 David Schleef <ds@schleef.org>
+2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac: Add sunaudio
- * examples/Makefile.am: make gstplay depend on gconf
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms
- * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette),
- (convert_table_lookup), (img_convert): remove c99-isms
- * gst/ffmpegcolorspace/imgconvert_template.h: make a constant
- unsigned, to fix a warning on Solaris
- * gst/mpeg1sys/systems.c: bcopy->memcpy
- * gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy
- * sys/Makefile.am: Add sunaudio
+ sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
+ (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
+ Avoid type checking in buffer casts.
+ Avoid caps copy in buffer_alloc when we can.
+ Use pad_peer_accept.
-2004-05-18 Wim Taymans <wim@fluendo.com>
+2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_init),
- (gst_ogg_mux_sinkconnect), (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_push_page),
- (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_loop):
- Fix an ugly memleak where the muxer didn't flush enough ogg
- pages. This also resulted in badly muxed ogg files.
+ gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tag.h:
+ Oops, make that 'Since: 0.10.9'.
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/asfdemux/asfheaders.c :
- * gst/asfdemux/asfheaders.h :
- * gst/asfdemux/gstasfdemux.c :
- - fix ASF_OBJ_PADDING guid
- - add 3 new object guids (language list, metadata,
- extended stream properties)
- - add a function to parse extended header objects
+ API: add GstTagImageType enum to describe images contained in image tags (#345641).
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
+ (gst_tag_image_type_get_type):
+ API: add GstTagImageType enum to describe images contained
+ in image tags (#345641).
-2004-05-18 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- remove leftover debugging g_print
+ gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ Fix warnings with gst-inspect: "buffers-min" property
+ should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
+ typo in property description.
-2004-05-17 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
+
+ gst/: Avoid unnecessary class cast check in class_init functions (#337747).
+ Original commit message from CVS:
+ Patch by: Cody Russell <bratsche at gnome org>
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_class_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
+ * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init):
+ * gst/volume/gstvolume.c: (gst_volume_class_init):
+ Avoid unnecessary class cast check in class_init
+ functions (#337747).
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- Fix for when the first format in a discont event is not a
- byte-based one. Should fix #137710.
+2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
+ (gst_text_overlay_video_chain):
+ g_markup_escape_text() REALLY doesn't like non-UTF8 input
+ and doesn't validate its input either (and neither did
+ textoverlay it seems). Let's do that then and fix #345206.
- * m4/a52.m4 : fix compilation with -Wall -Werror
- * m4/libfame.m4 : idem
- * m4/libmikmod.m4 : idem
+2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
+ (find_syncframe), (find_limits), (assign_value),
+ (count_burst_unit), (gst_multi_fd_sink_new_client),
+ (gst_multi_fd_sink_handle_client_write),
+ (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
+ (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
+ (gst_multi_fd_sink_change_state):
+ * gst/tcp/gstmultifdsink.h:
+ Added shiny new burst-on-connect methods.
+ Add properties to control the minimal amount of data queued.
+ Small cleanups.
+ API: bytes-min property
+ API: time-min property
+ API: buffers-min property
+ API: burst-unit property
+ API: burst-value property
+ API: add-full signal
+ * gst/tcp/gsttcp-marshal.list:
+ Added new marshaller code for the new signal.
+ * tests/check/elements/multifdsink.c: (GST_START_TEST),
+ (multifdsink_suite):
+ Added testcases for new burst methods.
+
+2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
-2004-05-17 Benjamin Otte <otte@gnome.org>
+ * gst-plugins-base.spec.in:
+ update for latest changes
+ Original commit message from CVS:
+ update for latest changes
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- signal the new tags before giving up the reference
+2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-05-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/theora/theoradec.c: Implement clipping for accurate seeking.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
+ Implement clipping for accurate seeking.
+ Closes #345225
- * ext/shout2/gstshout2.c:
- use application/ogg instead of application/x-ogg (patch by Patrick
- Guimond, fixes #142432)
- * sys/oss/gstosselement.c: (gst_osselement_reset),
- (gst_osselement_sync_parms):
- don't set fragment size unless specified (fixes #142493)
+2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
-2004-05-17 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
+ Original commit message from CVS:
+ Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
+ (gst_video_scale_transform):
+ Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
- * configure.ac : fix compilation of v4l2src with "-Wall -Werror"
- fixes #142664
+2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-17 Benjamin Otte <otte@gnome.org>
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- compute offsets correctly for internal buffers so timestamps are set
- correctly when we can't seek. Also handle cases where there are no
- offsets. (based on a patch by David Moore, fixes #142507)
+2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-17 Benjamin Otte <otte@gnome.org>
+ configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
+ Original commit message from CVS:
+ * configure.ac:
+ Fix --disable-extern (can't set conditionals conditionally,
+ #343602).
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- use correct variable when determining amount of data to skip so we
- don't skip into the void and segfault
+2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-16 Benjamin Otte <otte@gnome.org>
+ tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c: (test_reuse),
+ (audioresample_suite):
+ Add test case for bug #342789 fixed below.
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Hi, I'm a memleak
+2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-16 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * gst/asfdemux/gstasfdemux.c:
- - fix a mem leak and always propagate tags
- - add WMV3 to known video codecs (but no decoder yet)
- - replace "surplus data" at end of audio header for what
- it is : codec specific data
- - fix a typo
+ gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init), (gst_audioresample_init),
+ (audioresample_start), (audioresample_stop),
+ (gst_audioresample_set_property), (gst_audioresample_get_property):
+ Implement GstBaseTransform::start and ::stop so that audioresample
+ can clear its internal state properly and be reused insted of
+ causing non-negotiated errors with playbin under some circumstances
+ (#342789).
+ * tests/check/elements/audioresample.c: (setup_audioresample),
+ (cleanup_audioresample):
+ Need to set element state here so that ::start and ::stop are
+ called.
-2004-05-16 Arwed v. Merkatz <v.merkatz@gmx.net>
+2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
- reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
+ Parse extra data better, apparently it's right behind
+ the normal strf header size. Fixes #343500.
- * gst-libs/gst/audio/audioclock.c:
- Fix wrong return type (#142205).
+2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams):
+ If we fail to set the buffer_time and period_time alsa
+ parameters, post a warning and leave alsa select a
+ default instead of failing. Fixes #342085
- * ext/mad/gstmad.c: (gst_mad_class_init), (gst_mad_init):
- Ignore CRCs by default (fixes #142566).
+2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
+ Original commit message from CVS:
+ ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
- (gst_alsa_mixer_close), (gst_alsa_mixer_supported),
- (gst_alsa_mixer_build_list), (gst_alsa_mixer_free_list),
- (gst_alsa_mixer_change_state), (gst_alsa_mixer_list_tracks),
- (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
- (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record):
- Fix for cases where we fail to attach to a mixer.
+2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
+ out in the header file and shouldn't be listed in the docs.
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ Fix it so that it doesn't crash in the debug statement.
+
+2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/libs/: add remaining symbols into correct setions
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ add remaining symbols into correct setions
+ * gst-libs/gst/audio/gstringbuffer.c:
+ fix incomplete docs
+ * gst-libs/gst/audio/gstringbuffer.h:
+ comment out not yet implemented function
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ add short descriptions
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ fix return value docs
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ simplify debug logging
+ * gst-libs/gst/riff/riff-read.h:
+ sync function prototype and docs
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ remove left over symbol
+
+2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
- Don't touch events after not owning them anymore.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
- (gst_wavparse_fmt), (gst_wavparse_other),
- (gst_wavparse_handle_seek), (gst_wavparse_loop),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
- (gst_wavparse_srcpad_event):
- * gst/wavparse/gstwavparse.h:
- Add seeking, fix querying.
+ Use GST_PLUGIN_DOCS macro in configure.ac, add
+ Original commit message from CVS:
+ * autogen.sh:
+ * configure.ac:
+ * docs/Makefile.am:
+ Use GST_PLUGIN_DOCS macro in configure.ac, add
+ --enable-plugin-docs default to autogen.sh and use
+ ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
-2004-05-16 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- - process comments even if they don't end with \0\0
- g_convert would ignore them if present and works well without them
+ ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
+ (gst_ogg_demux_loop):
+ Combine GstFlowReturn from the source pads to give a
+ meaningfull result to the upstream peer or to stop the
+ processing task in case of errors.
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsa.c: (gst_alsa_caps), (gst_alsa_get_caps):
- simplify caps
+ gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c: (cb_probe):
+ Try GST_TAG_CODEC as fallback when extracting the
+ codec name; more debug info.
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- don't write to memory we might not write to - g_convert does that
- for us anyway (fixes #142613)
- (gst_asf_demux_audio_caps):
- comment out gst_util_dump_mem
+ ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
+ Original commit message from CVS:
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
+ Extract language tags from ogm subtitle streams, so that
+ the subtitle menu choices are labelled correctly in
+ Totem (fixes #344708).
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- compute correct expected timestamps after seek (broken since
- last commit)
- * ext/gdk_pixbuf/pixbufscale.c: (pixbufscale_init):
- rename element and debugging category to gdkpixbufscale
+ ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
+ Original commit message from CVS:
+ Patch by: Alessandro Decina <alessandro at nnva dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
+ (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
+ (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
+ (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
+ Fix various leaks. Fixes #343699.
+ Add x-smoke mime type.
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- add error checking to snd_pcm_delay and remove duplicate call to
- snd_pcm_delay that caused issues (see inline code comments)
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
- make more readable and fix return value when snd_pcm_delay fails
- (fixes #142586)
+ gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add IDs for 'bext' chunks (see #343837).
-2004-05-15 Jan Schmidt <thaytan@mad.scientisti.com>
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type),
- (gst_pixbufscale_get_type), (gst_pixbufscale_base_init),
- (gst_pixbufscale_class_init), (gst_pixbufscale_getcaps),
- (gst_pixbufscale_link), (gst_pixbufscale_init),
- (gst_pixbufscale_handle_src_event), (pixbufscale_scale),
- (gst_pixbufscale_chain), (gst_pixbufscale_set_property),
- (gst_pixbufscale_get_property), (pixbufscale_init):
- * ext/gdk_pixbuf/pixbufscale.h:
- Add these files I forgot earlier
+2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-05-15 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/gdk_pixbuf/Makefile.am:
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- Add new pixbufscale element to scale RGB video
- using gdk_pixbuf, because gdk_pixbuf does BILINEAR
- and HYPER interpolation correctly.
- * ext/theora/theoraenc.c: (theora_enc_chain),
- Discard buffer and return if explicit caps could not be set
- (theora_enc_get_property):
- Make _get return kbps for the bitrate consistent with
- the _set function.
+ gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/subparse/samiparse.c: (sami_context_pop_state),
+ (handle_start_font), (end_sami_element):
+ Honour font face tags in SAMI subtitles (#344503).
+2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ po/POTFILES.in: add missing files containing translatable strings
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ add missing files containing translatable strings
- * ext/libvisual/visual.c: (gst_visual_chain):
- add missing visual_audio_analyze
+2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-14 David Schleef <ds@schleef.org>
+ docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
+ Original commit message from CVS:
+ * docs/libs/tmpl/.cvsignore:
+ we don't want those *.sgml files in CVS either
- * ext/esd/esdsink.c: (gst_esdsink_chain): Fix crash when ESD
- is killed while we're playing.
- * gst/qtdemux/qtdemux.c: (qtdemux_parse): call
- gst_element_no_more_pads().
+2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-14 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ ignore more
+ Original commit message from CVS:
+ * docs/libs/.cvsignore:
+ * tests/check/elements/.cvsignore:
+ * tests/check/libs/.cvsignore:
+ ignore more
- * gst-libs/gst/riff/riff-read.c :
- - fix INFO tag extraction in RIFF/AVI files
- because gst_event_unref (event) also freed taglist
- - avoid a mem leak
+2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
- * gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
+ docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ also commiting the changed Makefile.am (added more libs to the
+ doc-build)
- * gst/auparse/gstauparse.c :
- - add code (commented for now) to support audio/x-adpcm on src pad
- (we have no decoder for those layout yet)
+2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/cdxaparse/gstcdxaparse.c :
- * gst/cdxaparse/gstcdxaparse.h :
- - partial rewrite using RiffRead (ripped iain's wavparse code)
+ docs/libs/: first batch of reordering things, add index & hierarchy
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ first batch of reordering things, add index & hierarchy
- * gst/rtp/gstrtpL16enc.c : typo
- * gst/rtp/gstrtpgsmenc.c : typo
+2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-13 Benjamin Otte <otte@gnome.org>
+ * common:
+ * ext/alsa/Makefile.am:
+ * ext/cdparanoia/Makefile.am:
+ * ext/gnomevfs/Makefile.am:
+ * ext/libvisual/Makefile.am:
+ * ext/ogg/Makefile.am:
+ * ext/pango/Makefile.am:
+ * ext/theora/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * sys/v4l/Makefile.am:
+ * sys/ximage/Makefile.am:
+ * sys/xvimage/Makefile.am:
+ further clean up build
+ Original commit message from CVS:
+ further clean up build
- * configure.ac:
- check for exact version of libvisual, it's not supposed to be
- API/ABI stable yet
-
-2004-05-13 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
- signal no-more-pads
-
-2004-05-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_src_convert)
- Report which format was used for GST_FORMAT_DEFAULT
- * gst/debug/Makefile.am:
- * gst/debug/gstdebug.c: (plugin_init):
- * gst/debug/progressreport.c: (gst_progressreport_base_init),
- (gst_progressreport_class_init), (gst_progressreport_init),
- (gst_progressreport_report), (gst_progressreport_set_property),
- (gst_progressreport_get_property), (gst_progressreport_chain),
- (gst_progressreport_plugin_init):
- Add progressreport element for testing.
-
-2004-05-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_change_state):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_playback_init):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_capture_init):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
- (gst_v4lsrc_grab_frame):
- add more debugging
- send a discont at start
-
-2004-05-12 Colin Walters <walters@redhat.com>
-
- * gst/asfdemux/gstasfdemux.c (gst_asf_demux_process_segment): Avoid
- inflooping if we can't find a chunk. Or in other words, don't blow
- chunks if we don't have a chunk to blow.
-
-2004-05-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/audiofile/gstafsrc.c: (gst_afsrc_get):
- Remove old debug output
- * ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
- (gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
- (gst_dvdec_set_property), (gst_dvdec_get_property):
- Change the quality setting to an enum, so it works from gst-launch
- Don't renegotiate a non-linked pad. Allows audio only decoding.
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
- (gst_deinterlace_link), (gst_deinterlace_init):
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
- (gst_videodrop_link):
- Some caps negotiation fixes
-
-2004-05-12 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/tarkin/gsttarkin.c :
- - Change RANK from NONE to PRIMARY
- * ext/gdk_pixbuf/gstgdkpixbuf.c :
- - Change RANK from NONE to MARGINAL
- * ext/divx/gstdivxenc.c :
- - Change RANK from PRIMARY to NONE (encoder/spider issue)
-
-2004-05-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisenc.c: (vorbis_granule_time_copy),
- (gst_vorbisenc_push_packet):
- copy a function that was added between 1.0 and 1.0.1 until we
- depend on worthwhile features of post-1.0
-
-2004-05-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- enable shout2 by default
- * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
- (gst_shout2send_base_init), (gst_shout2send_init),
- (gst_shout2send_connect), (gst_shout2send_change_state):
- * ext/shout2/gstshout2.h:
- make this work again. Based on a patch by Zaheer Abbas Merali (fixes
- #142262)
- * ext/theora/theora.c: (plugin_init):
- don't set rank on encoders
+ configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
+ Original commit message from CVS:
+ * configure.ac:
+ use GST_PKG_CHECK_MODULES, cleans up output
-2004-05-11 Jeremy Simon <jesimon@libertysurf.fr>
+2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Use codec_data property instead of flag1 and flag2 for wma
+ * common:
+ * win32/common/config.h:
+ update to cvs
+ Original commit message from CVS:
+ update to cvs
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/cdxaparse/gstcdxaparse.c :
- - Add mpegversion to CAPS to make it link
- - Rank is as GST_RANK_SECONDARY instead of NONE
- * gst/auparse/gstauparse.c :
- - Document all audio encoding we can encounter from Solaris 9
- headers and libsndfile information.
- - Increase max. rate from 48000 to 192000 (to match other elements)
- - Don't try to play junk data between header and samples
+ ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
+ Add support for burn:// URIs (#343385); const-ify things a bit,
+ use G_N_ELEMENTS instead of hard-coded array size.
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
- * ext/libvisual/visual.c: (gst_visual_getcaps):
- use the right caps depending on endianness (I hope)
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_plugin_init):
- use GST_RANK_NONE for all non-decoding elements or spider gets
- mighty confused
+ gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
+ Fix up broken entities before passing them to libxml *sigh*.
+ (#343303).
-2004-05-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- Fix some odd cases and fix BE metadata parsing of unicode16 text.
+ * ChangeLog:
+ * configure.ac:
+ back to trunk
+ Original commit message from CVS:
+ back to trunk
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+=== release 0.10.8 ===
- * gst/switch/gstswitch.c: (gst_switch_release_pad),
- (gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
- (gst_switch_loop), (gst_switch_get_type):
- whoever that was: DO NOT IMPORT PRIVATE SYMBOLS THAT ARE NOT IN
- HEADERS. Had to be said.
+2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-10 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ releasing 0.10.8
+ Original commit message from CVS:
+ releasing 0.10.8
- * configure.ac: Add prototype Dirac support.
- * ext/Makefile.am:
- * ext/dirac/Makefile.am:
- * ext/dirac/gstdirac.cc:
- * ext/dirac/gstdiracdec.cc:
+2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 0.10.7.2 prerelease
+ Original commit message from CVS:
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * win32/common/config.h:
+ 0.10.7.2 prerelease
- * gst/auparse/gstauparse.c: (gst_auparse_class_init),
- (gst_auparse_init), (gst_auparse_chain),
- (gst_auparse_change_state):
- Hack around spider. Remove me some day please.
+2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ move last template doc snippets to source code and delete them
+ Original commit message from CVS:
+ * docs/libs/tmpl/gstaudio.sgml:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/libs/tmpl/gstmixer.sgml:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * docs/libs/tmpl/gsttuner.sgml:
+ * docs/libs/tmpl/gstxoverlay.sgml:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/tuner.c:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ move last template doc snippets to source code and delete them
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Fix for some uninitialized variables in previous patch, also
- makes it work. Fixes #142286 while we're at it.
+2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ * gst/gdp/gstgdppay.c:
+ adapt to new api
+ Original commit message from CVS:
+ adapt to new api
+
+2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ configure.ac: enable building of GDP elements
+ Original commit message from CVS:
+ * configure.ac:
+ enable building of GDP elements
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
+ (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
+ (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
+ (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
+ (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
+ (gst_gdp_pay_change_state):
+ * gst/gdp/gstgdppay.h:
+ add version 1.0
+
+2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
+ Original commit message from CVS:
+ * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
+ (theora_parse_drain_queue):
+ Mark DELTA_UNIT on non-keyframes.
- * gst/auparse/gstauparse.c:
- fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
- only unsupported formats are ADPCM/CCITT G.72x
- reviewed by Ronald
- * gst-libs/gst/audio/audio.h: adds 24bit depth to PCM (x-raw-int)
+2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
+ (gst_ring_buffer_samples_done):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Document better the fact that latency_time and buffer_time are values
+ stored in microseconds, and not the usual GStreamer nanoseconds.
+ Change the variables (compatibly) that store them from GstClockTime
+ to guint64 to make it more clear that they're not storing clock times.
+ Also, remove the bogus property description that says the user can
+ specify -1 to get the default value, since that's never been the case.
+ When computing the default segment size for the ring buffer, make it
+ an integer number of samples.
+ When the sub-class indicates a delay greater than the number of
+ samples we've written return 0 from the audio sink get_time method.
+
+2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
+
+ tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (set_channel_positions),
+ (get_float_mc_caps), (get_int_mc_caps):
+ * tests/check/elements/audioresample.c:
+ * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
+ * tests/check/elements/videorate.c:
+ * tests/check/elements/videotestsrc.c: (GST_START_TEST):
+ * tests/check/elements/volume.c:
+ * tests/check/elements/vorbisdec.c:
+ * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
+ Don't busy-wait in tests; this was causing test timeouts very
+ frequently when running under valgrind.
-2004-05-10 Wim Taymans <wim@fluendo.com>
+2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/vorbis/Makefile.am:
- * ext/vorbis/README:
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_formats),
- (oggvorbisenc_get_type), (vorbis_caps_factory), (raw_caps_factory),
- (gst_oggvorbisenc_base_init), (gst_oggvorbisenc_class_init),
- (gst_oggvorbisenc_sinkconnect), (gst_oggvorbisenc_convert_src),
- (gst_oggvorbisenc_convert_sink),
- (gst_oggvorbisenc_get_query_types), (gst_oggvorbisenc_src_query),
- (gst_oggvorbisenc_init), (gst_oggvorbisenc_get_tag_value),
- (gst_oggvorbisenc_metadata_set1), (gst_oggvorbisenc_set_metadata),
- (get_constraints_string), (update_start_message),
- (gst_oggvorbisenc_setup), (gst_oggvorbisenc_write_page),
- (gst_oggvorbisenc_chain), (gst_oggvorbisenc_get_property),
- (gst_oggvorbisenc_set_property), (gst_oggvorbisenc_change_state):
- * ext/vorbis/oggvorbisenc.h:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisenc.c: (vorbis_caps_factory),
- (raw_caps_factory), (gst_vorbisenc_class_init),
- (gst_vorbisenc_init), (gst_vorbisenc_setup),
- (gst_vorbisenc_push_packet), (gst_vorbisenc_chain),
- (gst_vorbisenc_get_property), (gst_vorbisenc_set_property):
- * ext/vorbis/vorbisenc.h:
- Added a raw vorbis encoder to be used with the oggmuxer.
- We still need the old encoder for some gnome applications,
- read the README to find out how that works.
- The raw encoder is called "rawvorbisenc" until 0.9.
-
-2004-05-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init),
- (gst_ogg_print):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type),
- (gst_ogg_mux_base_init), (gst_ogg_mux_class_init),
- (gst_ogg_mux_get_sink_event_masks), (gst_ogg_mux_init),
- (gst_ogg_mux_sinkconnect), (gst_ogg_mux_pad_link),
- (gst_ogg_mux_pad_unlink), (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_handle_src_event), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_push_page), (gst_ogg_mux_compare_pads),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property),
- (gst_ogg_mux_change_state), (gst_ogg_mux_plugin_init):
- Added an ogg muxer.
- Small typo fixes in the demuxer.
-
-2004-05-10 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_push_packet), (theora_enc_chain),
- (theora_enc_change_state), (theora_enc_set_property),
- (theora_enc_get_property):
- Mark the last packet with an EOS flag which is not really needed
- in gstreamer.
- Do some better video framerate initialisation.
- Update the buffer timestamp.
-
-2004-05-10 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_change_state):
- Return the result of the parent state change call
-
-2004-05-10 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * gst/law/alaw.c : alawdec should be registered with type ALAWDEC, not ALAWENC
- * gst/law/alaw-decode.c : put audio/x-alaw on pads, instead of audio/x-mulaw
- * gst/law/alaw-encode.c : (idem)
- * ext/a52dec/gsta52dec.c : mark audio/a52, audio/ac3 as deprecated in a comment
- * gst/ac3parse/gstac3parse.c : audio/ac3 => audio/x-ac3
- * gst/realmedia/rmdemux.c : audio/a52 => audio/x-ac3
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- don't use a fixed buffer size when writing variable length data to
- it. Fixes memory corruption and makes alsasrc work
-
-2004-05-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.h:
+ small fixes
+ Original commit message from CVS:
+ small fixes
+
+2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * tests/check/elements/multifdsink.c:
+ fail_if_can_read is racy
+ Original commit message from CVS:
+ fail_if_can_read is racy
+
+2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/tcp/: make multifdsink properly deal with streamheader:
+ Original commit message from CVS:
+ * gst/tcp/README:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
+ (gst_multi_fd_sink_remove_client_link),
+ (gst_multi_fd_sink_client_queue_caps),
+ (gst_multi_fd_sink_client_queue_buffer),
+ (gst_multi_fd_sink_handle_client_write),
+ (gst_multi_fd_sink_render):
+ * gst/tcp/gstmultifdsink.h:
+ make multifdsink properly deal with streamheader:
+ - streamheader is taken from caps
+ - buffers marked with IN_CAPS are not sent
+ - streamheaders are sent, on connection, from the caps of the
+ buffer where the client gets positioned to
+ - further streamheader changes are done every time the client
+ will receive a buffer with different caps
+ * tests/check/elements/multifdsink.c: (GST_START_TEST),
+ (gst_multifdsink_create_streamheader):
+ add tests for this
- * ext/gnomevfs/gstgnomevfssink.c:
- (_gst_boolean_allow_overwrite_accumulator),
- (gst_gnomevfssink_class_init), (gst_gnomevfssink_open_file):
- Run glib's default signal handler (??) in RUN_CLEANUP rather than
- RUN_LAST, and don't use that to set the accumulator value because
- then it's always FALSE.
+2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
-2004-05-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
+ Reinstate limit on channel count. Vorbis does not define the meaning
+ of > 6 channels, so they're just independent channels. Gstreamer
+ currently has no mechanism to represent N independent channels.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
- Fix for unaligned RIFF files (i.e. where all the chunks together
- in a LIST chunk are not of the same size as the size given in
- the LIST chunk header). Fixes several odd WAVE files. Also fix
- ADPCM (block_align property) in audio, so that wavparse based
- on this works now as it used to stand-alone.
-
-2004-05-09 Edward Hervey <bilboed@bilboed.com>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/a52dec/gsta52dec.c:
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
- * ext/faac/gstfaac.c: (gst_faac_base_init):
- * ext/faad/gstfaad.c: (gst_faad_base_init):
- * ext/ivorbis/vorbisfile.c:
- * ext/lame/gstlame.c:
- * ext/libfame/gstlibfame.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
- * ext/sidplay/gstsiddec.cc:
- * ext/speex/gstspeexdec.c:
- * ext/speex/gstspeexenc.c:
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- correct klasses. Mostly s,Codec/(Audio|Video),\1/Codec,
- (fixes #142193)
-
-2004-05-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsa.c: (device_list),
- (gst_alsa_class_probe_devices):
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Fix alsa oddness in mixer after the combination of using mixer
- in source/sink elements and using hw:x,y instead of just hw:x.
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * gst/wavparse/gstwavparse.c: (gst_wavparse_destroy_sourcepad),
- (gst_wavparse_create_sourcepad):
- make PAUSED=>READY=>PAUSED=READY work by not destroying NULL
- sourcepads
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- allow discont events before caps nego
-
-2004-05-08 Benjamin Otte <otte@gnome.org>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- don't leak events
-
-2004-05-08 Benjamin Otte <otte@gnome.org>
-
- * gst/level/gstlevel.c: (gst_level_link), (gst_level_chain),
- (gst_level_change_state), (gst_level_init):
- * gst/level/gstlevel.h:
- figure out if we're initialized directly instead of keeping a
- variable that's wrong in 90% of cases
- don't initialize pads and then leak them and use a new unitialized
- pad. (fixes #142084)
- these were bugs so n00bish I didn't find them for an hour :/
-
-2004-05-08 Iain <iain@prettypeople.org>
-
- * gst/wavparse/gstwavparse.[ch]: Rewrote to use RiffRead instead.
- * gst-libs/gst/riff/riff-read.c (gst_riff_read_peek_head): Unstatic it
- (gst_riff_read_element_data): Ditto, and added a got_bytes argument to
- return the length that was read.
- (gst_riff_read_strf_auds): Allow fmt tags as well.
-
-2004-05-07 David Schleef <ds@schleef.org>
-
- * ext/faad/gstfaad.c: (gst_faad_sinkconnect): HACK to correct
- signed char assumption in faad.h.
-
-2004-05-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps):
- Missing break, detected by Daniel Gazard <daniel.gazard@free.fr>.
-
-2004-05-07 Colin Walters <walters@redhat.com>
-
- * gst/volume/gstvolume.c (gst_volume_dispose): Unref dpman.
- * ext/flac/gstflacdec.c (gst_flacdec_dispose): Add dispose
- function.
- * gst/audioscale/gstaudioscale.c (gst_audioscale_dispose):
- Add dispose function.
-
-2004-05-08 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/dv/gstdvdec.c: (gst_dvdec_video_link):
- Fix caps nego and pad templates. RGB mode caps should
- work now.
- * ext/dvdnav/gst-dvd:
- Move mpeg2dec inside the thread because otherwise the
- queue rejects cap changes mid-stream
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_get_type),
- (gst_mpeg2dec_flush_decoder):
- For mpeg2dec > 0.4.0, call the flush function instead of
- manually extracting all in-flight frames.
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_factory),
- (gst_dv1394src_init), (gst_dv1394src_iso_receive):
- Change mime type video/dv go video/x-dv to match the
- rest of gst-plugins
-
-2004-05-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_type),
- (gst_alsa_sink_class_init):
- * ext/alsa/gstalsasink.h:
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_get_type),
- (gst_alsa_src_class_init):
- * ext/alsa/gstalsasrc.h:
- Make alsasink/src a subclass of alsamixer so that mixer stuff
- shows up in gst-rec. Needs some finetuning.
+2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
-2004-05-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
+ Don't arbitrarily restrict channel counts and rate in vorbis.
+ In terms of effects likely on real-world files, this fixes 96kHz
+ playback of vorbis.
- * ext/lame/gstlame.c: (gst_lame_chain):
- simplify
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- fix event leak
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- be able to detect mp3 files < 4096 bytes
+2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ gst/audioconvert/audioconvert.c: More correct float->int conversion.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float):
+ More correct float->int conversion.
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_push_packet), (theora_enc_chain),
- (theora_enc_set_property), (theora_enc_get_property):
- Also encode the first frame, cleanup some code.
+2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
+ Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
+ value. Fixes g-critical on trying to play back ogg containing
+ unknown codec.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Forward events first before deciding that negotiation was
- not performed.
+2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
+ (setup_source):
+ * gst/playback/gstplaybasebin.h:
+ Make the subtitle detection work from any thread so we don't
+ deadlock. Fixes #343397.
+
+2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
+ (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
+ (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
+ (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
+ (gst_gdp_pay_get_property):
+ add crc-header and crc-payload properties
+ don't error out on some things that are recoverable
+ * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
+ add test for crc
+
+2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/gsttcp.c:
+ show type number when packet is of the wrong type
+ Original commit message from CVS:
+ show type number when packet is of the wrong type
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- First process the events before deciding that negotiation
- was not performed.
+2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
+ Original commit message from CVS:
+ * gst/volume/Makefile.am:
+ Seriously, it's not *that* hard to get compilation right. Even
+ a drunk can do it ! Add LIBOIL CFLAGS and LIBS
- * ext/theora/Makefile.am:
- * ext/theora/theora.c: (plugin_init):
- * ext/theora/theoradec.c: (theora_dec_change_state):
- * ext/theora/theoraenc.c: (gst_theora_enc_base_init),
- (gst_theora_enc_class_init), (gst_theora_enc_init),
- (theora_enc_sink_link), (theora_enc_event), (theora_push_packet),
- (theora_enc_chain), (theora_enc_change_state),
- (theora_enc_set_property), (theora_enc_get_property):
- Added a theora encoder, grouped the encoder and decoder into the
- same plugin.
-
-2004-05-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
- (gst_jpegenc_chain):
- fix DURATION on outgoing buffers
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
- debug using time formats
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_sink_link):
- windows with width/height 0 generate X errors, so don't allow them
-
-2004-05-05 Wim Taymans <wim@fluendo.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (src_templ),
- (gst_mpeg2dec_base_init), (gst_mpeg2dec_init),
- (gst_mpeg2dec_negotiate_format):
- * ext/mpeg2dec/gstmpeg2dec.h:
- removed the static pad template so that we can add the
- more accurate framerate value to the caps.
-
-
-2004-05-04 Benjamin Otte <otte@gnome.org>
+2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
+ Original commit message from CVS:
+ * ext/alsaspdif/alsaspdifsink.h:
+ * ext/amrwb/gstamrwbdec.h:
+ * ext/amrwb/gstamrwbenc.h:
+ * ext/amrwb/gstamrwbparse.h:
+ * ext/arts/gst_arts.h:
+ * ext/artsd/gstartsdsink.h:
+ * ext/audiofile/gstafparse.h:
+ * ext/audiofile/gstafsink.h:
+ * ext/audiofile/gstafsrc.h:
+ * ext/audioresample/gstaudioresample.h:
+ * ext/bz2/gstbz2dec.h:
+ * ext/bz2/gstbz2enc.h:
+ * ext/dirac/gstdiracdec.h:
+ * ext/directfb/dfbvideosink.h:
+ * ext/divx/gstdivxdec.h:
+ * ext/divx/gstdivxenc.h:
+ * ext/dts/gstdtsdec.h:
+ * ext/faac/gstfaac.h:
+ * ext/gsm/gstgsmdec.h:
+ * ext/gsm/gstgsmenc.h:
+ * ext/ivorbis/vorbisenc.h:
+ * ext/libfame/gstlibfame.h:
+ * ext/nas/nassink.h:
+ * ext/neon/gstneonhttpsrc.h:
+ * ext/polyp/polypsink.h:
+ * ext/sdl/sdlaudiosink.h:
+ * ext/sdl/sdlvideosink.h:
+ * ext/shout/gstshout.h:
+ * ext/snapshot/gstsnapshot.h:
+ * ext/sndfile/gstsf.h:
+ * ext/swfdec/gstswfdec.h:
+ * ext/tarkin/gsttarkindec.h:
+ * ext/tarkin/gsttarkinenc.h:
+ * ext/theora/theoradec.h:
+ * ext/wavpack/gstwavpackdec.h:
+ * ext/wavpack/gstwavpackparse.h:
+ * ext/xine/gstxine.h:
+ * ext/xvid/gstxviddec.h:
+ * ext/xvid/gstxvidenc.h:
+ * gst/cdxaparse/gstcdxaparse.h:
+ * gst/cdxaparse/gstcdxastrip.h:
+ * gst/colorspace/gstcolorspace.h:
+ * gst/festival/gstfestival.h:
+ * gst/freeze/gstfreeze.h:
+ * gst/gdp/gstgdpdepay.h:
+ * gst/gdp/gstgdppay.h:
+ * gst/modplug/gstmodplug.h:
+ * gst/mpeg1sys/gstmpeg1systemencode.h:
+ * gst/mpeg1videoparse/gstmp1videoparse.h:
+ * gst/mpeg2sub/gstmpeg2subt.h:
+ * gst/mpegaudioparse/gstmpegaudioparse.h:
+ * gst/multifilesink/gstmultifilesink.h:
+ * gst/overlay/gstoverlay.h:
+ * gst/playondemand/gstplayondemand.h:
+ * gst/qtdemux/qtdemux.h:
+ * gst/rtjpeg/gstrtjpegdec.h:
+ * gst/rtjpeg/gstrtjpegenc.h:
+ * gst/smooth/gstsmooth.h:
+ * gst/smoothwave/gstsmoothwave.h:
+ * gst/spectrum/gstspectrum.h:
+ * gst/speed/gstspeed.h:
+ * gst/stereo/gststereo.h:
+ * gst/switch/gstswitch.h:
+ * gst/tta/gstttadec.h:
+ * gst/tta/gstttaparse.h:
+ * gst/videodrop/gstvideodrop.h:
+ * gst/xingheader/gstxingmux.h:
+ * sys/directdraw/gstdirectdrawsink.h:
+ * sys/directsound/gstdirectsoundsink.h:
+ * sys/dxr3/dxr3audiosink.h:
+ * sys/dxr3/dxr3spusink.h:
+ * sys/dxr3/dxr3videosink.h:
+ * sys/qcam/gstqcamsrc.h:
+ * sys/vcd/vcdsrc.h:
+ Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
+
+2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func),
+ (volume_update_real_volume), (gst_volume_class_init),
+ (gst_volume_init), (volume_process_float), (volume_process_int16),
+ (volume_process_int16_clamp), (volume_set_caps),
+ (volume_transform_ip), (plugin_init):
+ * gst/volume/gstvolume.h:
+ rewrite the passthrough check, split _int16 and _int16_clamp, fix
+ another property desc., remove unused param from process function
+ * tests/check/elements/volume.c: (volume_suite):
+ reactivate the passthrough test
+
+2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixerelement.h:
+ * ext/alsa/gstalsamixeroptions.h:
+ * ext/alsa/gstalsamixertrack.h:
+ * ext/gnomevfs/gstgnomevfssink.h:
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/gsttheoraparse.h:
+ * ext/vorbis/vorbisparse.h:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ * gst/audioconvert/gstaudioconvert.h:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ * gst/playback/gststreamselector.h:
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcpclientsink.h:
+ * gst/tcp/gsttcpclientsrc.h:
+ * gst/tcp/gsttcpserversink.h:
+ * gst/tcp/gsttcpserversrc.h:
+ * gst/videorate/gstvideorate.h:
+ * gst/videoscale/gstvideoscale.h:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/volume/gstvolume.h:
+ * sys/v4l/gstv4ljpegsrc.h:
+ * sys/v4l/gstv4lmjpegsink.h:
+ * sys/v4l/gstv4lmjpegsrc.h:
+ * sys/v4l/gstv4lsrc.h:
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.h:
+ * tests/old/testsuite/alsa/sinesrc.h:
+ Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
+
+2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * tests/check/elements/multifdsink.c:
+ remove wrong commit
+ Original commit message from CVS:
+ remove wrong commit
+
+2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Handle DISCONT.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_reset),
+ (gst_visual_sink_setcaps), (gst_visual_sink_event),
+ (gst_visual_src_event), (get_buffer), (gst_visual_chain):
+ Handle DISCONT.
+ Use running time before doing QoS.
+ Handle mono too.
+
+2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ set a magic variable to indicate we know the docs are incomplete
- * configure.ac:
- check for kdemacros.h, too (should fix #141821)
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event), (vorbis_dec_chain):
- don't crash if no header was sent, but nicely error out (fixes part
- of #141554)
+2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-05-04 Wim Taymans <wim@fluendo.com>
+ win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
+ Original commit message from CVS:
+ * win32/common/libgstvideo.def:
+ export gst_video_calculate_display_ratio
+ * win32/vs6/libgstvideoscale.dsp:
+ add link to libgstvideo-0.10.lib
- * ext/mpeg2enc/gstmpeg2enc.cc: (gst_mpeg2enc_dispose): call the
- parent dispose function to avoid segfault on destroy.
+2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gen_source_element):
+ Throw a more comprehensible error for rtsp:// URIs (rather
+ than erroring out with a negotiation error later on) until
+ we fix playbin to handle rtspsrc etc.
- * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
- (plugin_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_sink_link):
- clean up debugging caps
- also recreate xvimage when format has changed
+2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/pango/gsttextoverlay.c: Added some FIXMEs.
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
+ (gst_text_overlay_text_event):
+ Added some FIXMEs.
+
+2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.*: Implement release_request_pad.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
+ (gst_adder_request_new_pad), (gst_adder_release_pad):
+ * gst/adder/gstadder.h:
+ Implement release_request_pad.
+ Make padcounter atomic.
+ * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
+ Added check for release_pad in adder.
+
+2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Fix build again.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
+ Fix build again.
+
+2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
+ (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
+ (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
+ (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
+ (gst_ogg_demux_bisect_forward_serialno),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
+ (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
+ add more debugging
+ clean up printf formats for granulepos and serialno
-2004-05-04 Benjamin Otte <otte@gnome.org>
+2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/libvisual/Makefile.am:
- * ext/libvisual/visual.c: (gst_visual_class_init),
- (gst_visual_init), (gst_visual_dispose), (gst_visual_getcaps),
- (gst_visual_srclink), (gst_visual_chain),
- (gst_visual_change_state), (plugin_init):
- use a GstAdapter to correctly adapt buffer sizes - allows using a
- framerate
-
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_fps), (gst_v4lsrc_getcaps),
- (gst_v4lsrc_buffer_free):
- * sys/v4l/v4l_calls.c: (gst_v4l_get_capabilities):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_grab_frame),
- (gst_v4lsrc_requeue_frame):
- move some debugging categories around
- query for fps index and set accordingly if found
-
-2004-05-03 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/lame/gstlame.c:
- correct defaults that lame_init puts out of range
-
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/divx/gstdivxenc.c: (gst_divxenc_get_type),
- (gst_divxenc_class_init):
- fix range since -1 is the default
- * gst/mpeg1sys/gstmpeg1systemencode.c:
- (gst_mpeg1_system_encode_get_type), (gst_system_encode_multiplex):
- * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_get_type),
- (gst_rtjpegdec_chain):
- * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_get_type),
- (gst_rtjpegenc_chain):
- * sys/qcam/gstqcamsrc.c: (gst_autoexp_mode_get_type),
- (gst_qcamsrc_get_type), (gst_qcamsrc_change_state):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_get_type):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get_type):
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_playback_init):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_capture_init):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init):
- * sys/vcd/vcdsrc.c: (vcdsrc_get_type), (vcdsrc_get):
- remove gst_info calls
+ * tests/check/elements/multifdsink.c:
+ * tests/check/generic/states.c:
+ properly fail if we can't make an element
+ Original commit message from CVS:
+ properly fail if we can't make an element
+
+2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (raw_caps_factory),
+ (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
+ (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
+ (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
+ (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
+ * ext/vorbis/vorbisenc.h:
+ Multi-channel caps negotiation, so we can do proper multichannel
+ vorbis encoding, negotiated through audioconvert.
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
- * Makefile.am:
- * po/af.po:
- * po/az.po:
- * po/en_GB.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- Updated translations
+ tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
+ Original commit message from CVS:
+ * tests/check/elements/adder.c: (test_event_message_received),
+ (test_play_twice_message_received), (GST_START_TEST),
+ (adder_suite):
+ Added check to show that #339935 is fixed with ongoing
+ adder and collectpads fixes.
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- refactor/comment code
+ gst/adder/gstadder.c: Don't leak pad name.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_request_new_pad):
+ Don't leak pad name.
-2004-05-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.c: Fix adder seeking.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_query_duration),
+ (forward_event_func), (forward_event), (gst_adder_src_event):
+ Fix adder seeking.
+ Make query/seeking code threadsafe.
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (test_event_message_received),
+ (GST_START_TEST), (test_play_twice_message_received):
+ Fix adder test case.
+
+2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_init), (gst_play_base_bin_dispose),
+ (set_encoding_element), (decodebin_element_added_cb),
+ (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
+ (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybasebin.h:
+ Add 'subtitle-encoding' property to playbin, so applications can
+ force a subtitle encoding for non-UTF8 subtitles (#342268).
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
+ (gst_sub_parse_set_property):
+ Rename recently-added 'encoding' property to 'subtitle-encoding'
+ (so it can be proxied by playbin/decodebin in a generic way
+ with less danger of false positives).
- * gst/asfdemux/Makefile.am:
- * gst/asfdemux/asfheaders.c:
- * gst/asfdemux/asfheaders.h:
- * gst/asfdemux/gstasf.c: (plugin_init):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_type),
- (gst_asf_demux_base_init), (gst_asf_demux_process_comment),
- (gst_asf_demux_setup_pad):
- * gst/asfdemux/gstasfdemux.h:
- * gst/asfdemux/gstasfmux.c:
- * gst/asfdemux/gstasfmux.h:
- Add tagging support to demuxer, split out registration in its own
- file instead of in demux (hacky), and prevent having some tables
- in our memory multiple times (in asfheaders.h).
+2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
-2004-05-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
+ (append_with_other_format), (set_structure_widths),
+ (gst_audio_convert_transform_caps):
+ Patch from #341562: give more specific audio caps in get_caps, so
+ that basetransform can make better decisions on what caps to
+ negotiate.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_metadata):
- * gst/matroska/matroska-ids.h:
- Basic tag reading support.
+2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/elements/volume.c: make it compile again
+ Original commit message from CVS:
+ * tests/check/elements/volume.c:
+ make it compile again
+
+2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/elements/volume.c: disable test until #343196 gets resolved
+ Original commit message from CVS:
+ * tests/check/elements/volume.c: (volume_suite):
+ disable test until #343196 gets resolved
+
+2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/adder/gstadder.c: Make it easier to copy&paste
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_get_type):
+ Make it easier to copy&paste
+ * gst/volume/Makefile.am:
+ * gst/volume/gstvolume.c: (volume_update_real_volume),
+ (gst_volume_set_volume), (gst_volume_set_mute),
+ (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
+ (volume_transform_ip), (volume_update_mute),
+ (volume_update_volume):
+ * gst/volume/gstvolume.h:
+ Add own debug category, move duplicate code to helper function, fix
+ property texts, add more comments and prepare ffor liboil-goodness
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
+ add test for mute and passtrough case, be a bit more verbose to track
+ failure
+ * tests/check/generic/states.c: (GST_START_TEST):
+ catch elements that fail to instantiate
+
+2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
+ Original commit message from CVS:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/check/pipelines/theoraenc.c:
+ * tests/check/pipelines/vorbisenc.c:
+ Comment out tests using parse_launch() if core was built without
+ parsing capabilities.
+
+2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Extra bonus points for whoever explains to ensonic that you are meant
+ to test unit tests thoroughly before commiting them, especially if
+ you know it's going to break.
+ De-activated element/adder tests.
+
+2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
+ Marking caps conversion issues as GST_WARNING is way too verbose,
+ Moving them to GST_LOG.
+
+2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
+ Original commit message from CVS:
+ * README:
+ Replace current README (containing the release notes from
+ some 0.9.x version) with a proper README taken from the core.
-2004-04-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
- Really detect ac-3 audio.
- * gst/typefind/gsttypefindfunctions.c: (matroska_type_find):
- really detect matroska files (off-by-1).
+ ext/vorbis/vorbisdec.c: Small cleanups.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
+ (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
+ (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
+ (vorbis_dec_change_state):
+ Small cleanups.
+ Add some FIXMEs
+ Clip output samples to segment boundaries.
-2004-04-30 David Schleef <ds@schleef.org>
+2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_type_get),
- (qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_dump_co64),
- (qtdemux_dump_unknown), (qtdemux_parse_tree), (qtdemux_parse_udta),
- (qtdemux_tag_add), (get_size), (gst_qtdemux_handle_esds): More qtdemux
- hackage -- parse a lot more atoms, extract a few tags. One might even
- mistake this for tag support. Maybe it is.
- * gst/qtdemux/qtdemux.h:
+ sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
+ (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
+ Improve the errors produced on bad output, including some human
+ readable description strings.
+ Handle the (theoretical for ximagesink) case where the XServer
+ has a different idea about the size required for a particular
+ frame and gives us too small a memory allocation.
-2004-04-30 Colin Walters <walters@verbum.org>
+2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsasink.c (gst_alsa_sink_mmap): Plug a memleak.
+ * ChangeLog:
+ Mention bugs fixed by previous commit
+ Original commit message from CVS:
+ Mention bugs fixed by previous commit
+
+2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
+ (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
+ Improve the errors produced on bad output, including some human
+ readable description strings.
+ Handle RGB Xv formats properly by transforming them into our
+ big-endian caps description.
+ Use gst_caps_truncate to ensure that we never try and choose a
+ non-fixed caps in buffer_alloc.
+ Handle the case where the XServer has a different idea about the size
+ required for a particular frame and gives us too small a memory
+ allocation.
+ Use -1 to indicate 'no image format', because 0 is a valid XServer
+ image format number.
+ Put RGB Xv formats at the end of the caps, so that we always prefer
+ YUV format frames.
+ Iterate the available Xv Encodings to determine the maximum width and
+ height, and then return that in our caps.
+
+2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
+ When there is only one unfinished pad and it receives an event that
+ doesn't match our requirements, we need to set alldone=FALSE so that
+ the fakesink is not removed yet.
+
+2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
+ Use gst_type_find_helper_for_buffer() to find the type
+ of stream from the first packet.
+ * configure.ac:
+ Bump requirements to core CVS (needed for vorbis
+ typefinding to work).
-2004-04-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_getcaps):
- remove broken nego fix
+ gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
+ Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
+ Else they play perfectly fine with qtdemux.
-2004-04-30 Benjamin Otte <otte@gnome.org>
+2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- * ext/Makefile.am:
- * ext/libvisual/Makefile.am:
- * ext/libvisual/visual.c:
- add initial support for libvisual (http://libvisual.sourceforge.net)
- libvisual is still quite alpha, so expect crashes in there :)
+ make more debug catagories static
+ Original commit message from CVS:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
+ * gst/audiorate/gstaudiorate.c:
+ make more debug catagories static
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (message_received),
+ (test_event_message_received), (GST_START_TEST),
+ (test_play_twice_message_received), (adder_suite):
+ added test case for using element twice, extra bonus points for anyone
+ who can make these test run reliably
-2004-04-29 David Schleef <ds@schleef.org>
+2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse),
- (qtdemux_parse_trak), (get_size), (gst_qtdemux_handle_esds): Hacked
- up qtdemux to make it spit out codec_data. Do _not_ look at this
- code; you will no longer respect me.
+ ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_chain):
+ Make work with time-stamped input buffers that do not
+ have a granulepos in BUFFER_OFFSET_END (like theora
+ buffers coming from matroskademux). Fixes #342448.
-2004-04-29 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
+ (gst_gdp_depay_change_state):
+ * gst/gdp/gstgdpdepay.h:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
+ (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
+ (gst_gdp_pay_change_state):
+ * gst/gdp/gstgdppay.h:
+ Handle error cases when calling functions
+ do downwards state change after parent's change_state
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/elements/gdppay.c: (GST_START_TEST):
+ clean up more
+
+2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
+ Original commit message from CVS:
+ * gst/gdp/Makefile.am:
+ * gst/gdp/gstgdp.c: (plugin_init):
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
+ (gst_gdp_depay_class_init), (gst_gdp_depay_init),
+ (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
+ (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
+ * gst/gdp/gstgdpdepay.h:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
+ (gst_gdp_pay_class_init), (gst_gdp_pay_init),
+ (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
+ (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
+ (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
+ (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
+ (gst_gdp_pay_plugin_init):
+ * gst/gdp/gstgdppay.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
+ (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
+ (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
+ * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
+ (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
+ (main):
+ adding GDP payloader and depayloader. Build integration will
+ follow later when the GDP issues for core are sorted out.
- * ext/alsa/gstalsa.c : (gst_alsa_class_probe_devices)
- * ext/alsa/gstalsa.h :
- change alsa pcm device discovery to find more than 1 device
- per card. code review by Ronald.
+2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
-2004-04-29 David Schleef <ds@schleef.org>
+ gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/tcp/Makefile.am:
+ fdstresstest doesn't need Gtk+, fix compilation if
+ gtk is not available (#342566).
- * sys/oss/gstosselement.c: (gst_osselement_rate_probe_check):
- Add a check for a driver bug on FreeBSD. (bug #140565)
+2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ 80 line columns
+ Removed redundant floor()
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_get_type):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
- (gst_jpegenc_getcaps):
- move format setting to inner loop
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_getcaps):
- use GST_PAD_CAPS if available so that we use already negotiated
- caps
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse_moov), (qtdemux_parse):
- extra debugging
- * sys/qcam/qcam-Linux.c: (qc_lock_wait), (qc_unlock):
- * sys/qcam/qcam-os.c: (qc_lock_wait), (qc_unlock):
- move hardcoded path to DEFINE
+2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-28 David Schleef <ds@schleef.org>
+ gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ On second thought, just skip JUNK chunks automatically, so
+ the caller doesn't have to handle this. Fixes #342345.
+ Also, return GST_FLOW_UNEXPECTED if we get a short read,
+ not GST_FLOW_ERROR.
- * gst/speed/gstspeed.c: (speed_parse_caps): Fix caps parsing.
- (bug #140064)
+2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ Don't bail out on JUNK chunks with a size of 0 (would try to
+ pull_range 0 bytes before, which sources don't like too much).
+ See #342345.
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
- Don't probe for playback device if we're a source element. Fixes
- #139658.
+2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
+ Use the gstutil scaling function to preserve 64 bits while calculating
+ output width and height from the display-aspect-ratio. (A continuation
+ of #341542)
+
+2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
+ (gst_xvimagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.h:
+ When performing buffer allocations, remember the caps and image format
+ we return so that if the same caps are asked for next time we can
+ return them immediately without doing any caps intersections.
+
+2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/README: Some new documentation
+ Original commit message from CVS:
+ 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst-libs/gst/rtp/README:
+ Some new documentation
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ New RTP audio base payloader class. Supports frame or sample based codecs.
+ Not enabled in Makefile.am until approved.
+
+2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
+ Original commit message from CVS:
+ * tests/check/elements/alsa.c: (test_device_property_probe):
+ Fix test case: don't try to free NULL GValueArray when there
+ are no devices.
-2004-04-29 Benjamin Otte <otte@gnome.org>
+2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_chain):
- rewrite buffer offset
+ tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/alsa.c: (test_device_property_probe),
+ (alsa_suite), (main):
+ Add simple test that runs a device property probe on alsasrc,
+ alsasink and alsamixer. Disable valgrind check for now (too
+ many leaks in libasound, and valgrind ignored my suppressions
+ additions).
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
+
+ ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
+ Original commit message from CVS:
+ * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
+ (gst_alsa_device_property_probe_probe_property),
+ (gst_alsa_device_property_probe_needs_probe),
+ (gst_alsa_device_property_probe_get_values),
+ (gst_alsa_type_add_device_property_probe_interface):
+ * ext/alsa/gstalsadeviceprobe.h:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_init_interfaces):
+ * ext/alsa/gstalsamixerelement.h:
+ Clean up and simplify alsa device probing. Make it actually work
+ for multiple classes. Don't cache results any longer.
+ * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
+ (gst_alsasink_init):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
+ (gst_alsasrc_interface_supported), (gst_implements_interface_init),
+ (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
+ Make alsasink and alsasrc implement the GstPropertyProbe interface
+ for device probing (#342181).
+ Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
- * configure.ac:
- * ext/Makefile.am:
- * ext/dts/Makefile.am:
- * ext/dts/gstdtsdec.c: (gst_dtsdec_get_type),
- (gst_dtsdec_base_init), (gst_dtsdec_class_init), (gst_dtsdec_init),
- (gst_dtsdec_channels), (gst_dtsdec_renegotiate),
- (gst_dtsdec_handle_event), (gst_dtsdec_update_streaminfo),
- (gst_dtsdec_loop), (gst_dtsdec_change_state),
- (gst_dtsdec_set_property), (gst_dtsdec_get_property),
- (plugin_init):
- * ext/dts/gstdtsdec.h:
- New DTS decoder.
- * ext/faad/gstfaad.c: (gst_faad_sinkconnect),
- (gst_faad_srcconnect):
- Add ESDS atom handling (.m4a).
-
-2004-04-27 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c: (plugin_init):
- Remove comment that makes no sense.
- * ext/mad/gstid3tag.c: (gst_id3_tag_set_property):
- Fix for obvious typo that resulted in warnings during gst-register.
- * ext/xvid/gstxviddec.c: (gst_xviddec_src_link),
- (gst_xviddec_sink_link):
- Fix caps negotiation a bit better.
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
- We call this 'codec_data', not 'esds'.
-
-2004-04-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/monoscope/gstmonoscope.c:
- make sure we only provide 256x128
- * gst/monoscope/monoscope.c: (monoscope_init):
- assert size of 256x128
-
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Makefile.am:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_buffer_free):
- fixate to max width and height of device
+ gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
+ Original commit message from CVS:
+ * gst/subparse/samiparse.c: (handle_start_font):
+ Don't ignore return value of strtol (++compiler_happiness).
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
- * Makefile.am:
- * sys/v4l/gstv4l.c:
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/v4l_calls.c:
- * sys/v4l/v4lsrc_calls.c:
- fix for qc-usb driver which fakes having more than one buffer
- by handing the same buffer twice, which confused GStreamer's/v4lsrc
- buffer_free override
- add debugging
+ gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist chollian net>
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_class_init), (gst_sub_parse_init),
+ (gst_sub_parse_set_property), (gst_sub_parse_get_property),
+ (convert_encoding):
+ * gst/subparse/gstsubparse.h:
+ Add 'encoding' property (#341681).
+ * gst/subparse/samiparse.c: (characters_sami):
+ Output is pango markup, so we need to escape text
+ between tags (#342143).
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Makefile.am:
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_change_state),
- (gst_videotestsrc_init), (gst_videotestsrc_get),
- (gst_videotestsrc_set_property), (gst_videotestsrc_get_property):
- * gst/videotestsrc/gstvideotestsrc.h:
- add num-buffers property
+ gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_check_channel_positions):
+ It's okay to have caps with channels=1 and a channel position
+ different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
+ (deinterleavers might want to keep the position in the caps,
+ so that they can be re-interleaved again properly later).
+ Leave check for unexpected 2-channel layouts intact for now.
- 2004-04-26 Benjamin Otte <otte@gnome.org>
+2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
+
+ gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
+ Original commit message from CVS:
+ 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
+ Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
+ basesrc can do its job correctly.
+
+2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
+ Original commit message from CVS:
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
+ (gst_alsa_detect_formats), (get_channel_free_structure),
+ (caps_add_channel_configuration), (gst_alsa_detect_channels),
+ (gst_alsa_probe_supported_formats):
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
+ Refactor and improve caps probing code: probe signedness
+ when we probe the supported formats/widths; set endianness
+ to the one we actually probed for (ie. cpu endianness).
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
+ (gst_alsasrc_close):
+ * ext/alsa/gstalsasrc.h:
+ Implement caps probing for alsasrc.
- * ext/mad/gstid3tag.c: (plugin_init):
- set id3mux rank to NONE so it doesn't confuse spider
- require audio/mpeg,mpegversion=1 in id3mux
+2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (gst_theora_dec_reset),
+ (theora_dec_src_query), (theora_dec_src_event),
+ (theora_dec_sink_event), (theora_handle_comment_packet),
+ (theora_handle_data_packet), (theora_dec_change_state):
+ Cleanups, add some G_LIKELY.
+ Use segment helpers instead of our own wrong code.
+ Clear queued buffers on seek and READY.
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
+ (vorbis_dec_convert), (vorbis_dec_src_query),
+ (vorbis_dec_src_event), (vorbis_dec_sink_event),
+ (vorbis_handle_comment_packet), (vorbis_dec_push),
+ (vorbis_handle_data_packet), (vorbis_dec_chain),
+ (vorbis_dec_change_state):
+ * ext/vorbis/vorbisdec.h:
+ Remove old useless packetno variable.
+ Do position query properly.
+ Add some G_LIKELY.
+ Do cleanup of queued buffers in new helper function
+ and use it.
+
+2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
+ Query supported sample rates. Fixes #341732.
-2004-04-26 Benjamin Otte <otte@gnome.org>
+2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
+ Original commit message from CVS:
+ 2006-05-15 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstdecodebin.c: (cleanup_decodebin),
+ (gst_decode_bin_change_state): Make decodebin reusable
+ when going from PAUSE_TO_READY and then back to PAUSED.
+ Fixes #331678.
+
+2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
+ (vorbis_dec_convert), (vorbis_dec_src_query),
+ (vorbis_dec_sink_query), (vorbis_dec_src_event),
+ (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
+ (vorbis_dec_clean_queued), (vorbis_dec_push),
+ (vorbis_handle_data_packet), (vorbis_dec_change_state):
+ Cleanups. Use refcounting and DEBUG_OBJECT.
+ Reset segment on flush, use code methods instead of our
+ own wrong version.
+ Fix potential memleak.
+
+2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
+ (gst_alsasink_init):
+ * ext/alsa/gstalsasink.h:
+ Don't leak allocated snd_output_t structure if there's
+ more than one alsasink instance at a time (#341873).
+ Also fix GObject macros in header file.
- * configure.ac:
- detect faad correctly as non-working if it's indeed non-working
+2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect):
+ Don't use libxml functions in the typefinding code.
- * Makefile.am:
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
- (gst_jpegenc_class_init), (gst_jpegenc_getcaps):
- fix _getcaps so it only negotiates to its supported format
+2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-04-25 Benjamin Otte <otte@gnome.org>
+ ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
+ Fix seeking performance in the case where a non-header
+ packet has a 0 granulepos (busted theora case).
+ Fixes #341719
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- fix memleak
+2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-23 Benjamin Otte <otte@gnome.org>
+ gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect):
+ Improve SAMI typefinding: handle case where there are
+ whitespaces or newlines in front of the first <SAMI>
+ tag (#169936).
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- audio/x-raw-int with height rules! not. Now it's depth.
+2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
+ Original commit message from CVS:
+ * configure.ac:
+ Build video4linux plugin even if there's no XVIDEO, just
+ without implementing the GstXOverlay interface (#334002).
- * gst/wavparse/gstwavparse.c: (gst_wavparse_create_sourcepad),
- (gst_wavparse_parse_fmt), (gst_wavparse_handle_sink_event),
- (gst_wavparse_loop):
- Missing variable initialization. Add handling of DVI ADPCM. Fix
- mis-parsing of LIST chunks. This works around a bug where we mis-
- parse non-aligning LIST chunks (so LIST chunks where the contents
- don't align with the actual LIST size). The correct fix is to use
- rifflib, I'm not going to fix wavparse - too much work. All this
- fixes #104878.
+2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Add tentative support for libvisual-0.4 (#336881).
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
+ (plugin_init):
+ Add tentative support for libvisual-0.4 (#336881).
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
- * ext/shout/gstshout.c: (gst_icecastsend_change_state):
- fix shoutcast not working (fixes #140844)
+ gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/subparse/samiparse.c: (handle_start_font):
+ Need to map "silver" colour explicitly (#169936).
-2004-04-22 Benjamin Otte <otte@gnome.org>
+2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ gst/subparse/: Add support for SAMI subtitles (#169936).
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
+ (gst_sub_parse_format_autodetect), (feed_textbuf),
+ (gst_subparse_type_find), (plugin_init):
+ * gst/subparse/gstsubparse.h:
+ * gst/subparse/samiparse.c:
+ * gst/subparse/samiparse.h:
+ Add support for SAMI subtitles (#169936).
- * ext/hermes/gsthermescolorspace.c:
- (gst_hermes_colorspace_caps_remove_format_info):
- * gst/colorspace/gstcolorspace.c:
- (gst_colorspace_caps_remove_format_info):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_caps_remove_format_info):
- s/gst_caps_simplify/gst_caps_do_simplify/
+2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-22 Benjamin Otte <otte@gnome.org>
+ * win32/common/config.h:
+ update config.h
+ Original commit message from CVS:
+ update config.h
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- mpegversion is an int
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
- don't try to create pad templates with NULL caps, use any caps
- instead.
+2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 David Schleef <ds@schleef.org>
+ * ext/ogg/README:
+ fix mistakes in README
+ Original commit message from CVS:
+ fix mistakes in README
- * ext/sdl/Makefile.am: Link against libgstinterfaces, not
- libgstxoverlay. jmmv@menta.net (Julio M. Merino Vidal)
- (bug #140384)
+2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
+ gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
+ Original commit message from CVS:
+ * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
+ Fix #341696: crash when mixing L+R+C to mono or stereo.
+ * tests/check/Makefile.am:
+ * tests/check/elements/audioconvert.c: (set_channel_positions),
+ (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
+ (audioconvert_suite):
+ Add test for the above, including some generic framework bits for
+ testing multichannel things.
- reviewed by David Schleef
+2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/mad/gstid3tag.c: Add stdlib.h
- * gst/rtp/gstrtpgsmenc.c: same
- * gst/tags/gstid3tag.c: same
- * gst/udp/gstudpsrc.c: (gst_udpsrc_get): Fix GST_DISABLE_LOADSAVE
- * gst/tcp/gsttcpsink.c: (gst_tcpsink_sink_link): Adjust
- GST_DISABLE_LOADSAVE use.
- * gst/udp/gstudpsink.c: (gst_udpsink_sink_link): Likewise.
- * gst/tcp/gsttcpsrc.c: (gst_tcpsrc_get): Likewise.
- * ext/gnomevfs/gstgnomevfssrc.c: Include <stdlib.h> (needed by
- atol(3)).
- * sys/oss/gstosselement.h: Include <sys/types.h> (needed for dev_t).
- * gst/tags/gstvorbistag.c: Include <stdlib.h> (needed by
- strtoul(3)).
- * gst/rtp/gstrtpL16enc.c: Include <stdlib.h> (needed by random(3)).
- * ext/mad/Makefile.am: (libgstmad_la_CFLAGS): Add $(MAD_CFLAGS)
- $(ID3_CFLAGS).
- * ext/libfame/Makefile.am: (libgstlibfame_la_CFLAGS): Add
- $(LIBFAME_CFLAGS).
+ * configure.ac:
+ Back to CVS
+ Original commit message from CVS:
+ Back to CVS
-2004-04-20 David Schleef <ds@schleef.org>
+=== release 0.10.7 ===
- * gst/realmedia/rmdemux.c: This was supposed to part of the
- last checkin. Same idea.
+2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
+ configure.ac: releasing 0.10.7, "Leave the gun"
+ Original commit message from CVS:
+ 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ releasing 0.10.7, "Leave the gun"
- reviewed by David Schleef
+2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac: bump required gstreamer version to 0.8.1.1
- because of following changes [--ds]
+ * common:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
- * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h.
- (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list)
- (gst_riff_read_header): Use GST_READ_UINT*
- macros to access possibly unaligned memory.
+2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h.
- (mp3_type_find): Use GST_READ_UINT*
- macros to access possibly unaligned memory.
- (mp3_type_find, mpeg1_parse_header, qt_type_find)
- (speex_type_find): Likewise
+ Fix the build.
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
+ Fix the build.
- * gst/tags/gstvorbistag.c: (ADVANCE): Likewise
+2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
+ * gst-libs/gst/video/video.h:
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
+ * tests/check/Makefile.am:
+ * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
+ (main):
+ Fix integer overflow problem with pixel-aspect-ratio calculations
+ in videoscale and xvimagesink (#341542)
- * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc).
- (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET)
- (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET)
- (gst_qtdemux_loop_header, gst_qtdemux_loop_header)
- (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type)
- (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT*
- macros to access possibly unaligned memory.
+2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk):
- Likewise.
+ gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead)
- (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise.
+2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Likewise.
+ win32/MANIFEST: update win32 files listing
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ update win32 files listing
- * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA)
- (gst_mpeg2subt_chain_subtitle): Likewise.
+2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq)
- (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain):
- Likewise.
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info):
- Likewise.
+2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes):
- Likewise.
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint):
- Likewise.
+2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
- * configure.ac:
- update required version of GStreamer because of GST_TIME_FORMAT
+2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
- * ext/mad/gstid3tag.c: (gst_id3_tag_init):
- remove leftover g_print
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- don't try setting only a subset of the caps. We don't want to kill
- autoplugging on purpose
+2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
+ Original commit message from CVS:
+ 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ patch by: Sjoerd Simons (sjoerd@luon.net)
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (group_create), (group_destroy), (add_stream),
+ (gst_play_base_bin_get_property),
+ (gst_play_base_bin_get_streaminfo_value_array):
+ * gst/playback/gstplaybasebin.h:
+ API: GstPlayBaseBin::stream-info-value-array property
+ use a more bindings-friendly way of exposing streaminfo
+ using a GValueArray. Tested in ipython.
+ Closes #341114
- * sys/ximage/ximagesink.c: (plugin_init):
- * sys/xvimage/xvimagesink.c: (plugin_init):
- add debugging categories
+2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * tests/check/elements/multifdsink.c:
+ fix some type warnings
+ Original commit message from CVS:
+ fix some type warnings
+
+2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
+ (queue_underrun_cb), (queue_filled_cb):
+ Also catch queue underruns but don't do anything yet.
+ Refactor and comment queue enlarging code a bit.
+ * gst/playback/gstplaybasebin.c: (queue_overrun),
+ (queue_threshold_reached), (queue_out_of_data),
+ (gen_preroll_element):
+ If a queue over/underruns check that we don't create nasty
+ deadlocks when the min-threshold is not reached but the
+ max-bytes is. In those cases disable max-bytes when we
+ know that the queue is fed timed data.
+ Add more comments.
+
+2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_audio_element):
+ Make playbin automatically plug an 'audioresample'
+ element before the audio sink as well. This solves
+ problems with sinks that only accept a very specific
+ sample rate, like esdsink (e.g. #340379).
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/en_GB.po:
- * po/LINGUAS:
- Adding en_GB translation (Gareth Owen)
-
-2004-04-20 David Schleef <ds@schleef.org>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (qtdemux_parse), (qtdemux_type_get), (qtdemux_dump_mvhd),
- (qtdemux_dump_tkhd), (qtdemux_dump_stsd), (qtdemux_dump_unknown),
- (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
- A number of new features and hacks to extract the esds atom and
- put it into the caps. (bug #137724)
-
-2004-04-19 David Schleef <ds@schleef.org>
-
- * gconf/Makefile.am: Fix for non-GNU make
- * gst-libs/gst/Makefile.am: Change directory order to handle
- GstPlay linking with gstinterfaces
- * gst-libs/gst/audio/make_filter: make use of tr portable
- * gst-libs/gst/play/Makefile.am: Add intended \
- * gst-libs/gst/xwindowlistener/xwindowlistener.c:
- (gst_xwin_set_clips): Switch to ISO variadic macro. Use a
- function prototype instead of void *.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c: Switch to ISO variadic
- macro.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain): wrap NULL in GST_ELEMENT_ERROR call
- * gst/videofilter/make_filter: make use of tr portable
- * pkgconfig/Makefile.am: Remove GNU extension in Makefile target
+ gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gen_source_element):
+ Make http sources send special headers so that we receive
+ icecast metadata if the http stream is an icecast stream
+ (otherwise the server will just ignore them). This also
+ means that from now on users will need the 'icydemux'
+ element from gst-plugins-good installed if they want to
+ listen to icecast radio streams. (#341432, #333657).
-2004-04-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * po/LINGUAS:
- * po/uk.po:
- Added Ukrainian translation (Maxim V. Dziumanenko)
+ * gst/tcp/gstmultifdsink.c:
+ more commenting
+ Original commit message from CVS:
+ more commenting
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gsm/gstgsmdec.c: (gst_gsmdec_init), (gst_gsmdec_getcaps),
- (gst_gsmdec_link), (gst_gsmdec_chain):
- Fix capsnego, simplify chain function slightly.
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Add GSM.
+ gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
+ (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
+ remove stupid example from docs - it should come with a simple
+ C program instead.
+ Clean up/fix docs
+ * tests/check/elements/multifdsink.c: (wait_bytes_served),
+ (fail_if_can_read), (GST_START_TEST),
+ (gst_multifdsink_create_streamheader), (multifdsink_suite):
+ add a test for changing streamheader which exposes a bug in
+ multifdsink
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_destroy_sourcepad), (gst_wavparse_create_sourcepad),
- (gst_wavparse_parse_fmt), (gst_wavparse_change_state):
- Hack to make wavparse work with spider (always -> sometimes pad).
- Fixes #135862 && #140411.
+ ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
+ (gst_gnome_vfs_src_received_headers_callback):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ Don't set icy-caps unless we have a sane interval value. Move
+ interval to a local variable; we never use it outside this function.
-2004-04-18 Benjamin Otte <otte@gnome.org>
+2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms),
- (gst_osselement_rate_probe_check),
- (gst_osselement_rate_check_rate), (gst_osselement_rate_add_rate):
- get rid of \n in debug output
+ sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
+ Register special buffer types along with the objects so
+ that they are not registered at runtime from N different
+ streaming threads since they are not threadsafe.
-2004-04-17 Iain <iain@prettypeople.org>
+2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/wavparse/gstwavparse.c (gst_wavparse_loop): Allow all events,
- not just EOS.
+ * tests/check/elements/multifdsink.c:
+ set caps and plug leaks
+ Original commit message from CVS:
+ set caps and plug leaks
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstid3tag.c: (gst_id3_tag_get_type),
- (gst_id3_tag_class_init), (gst_id3_tag_get_caps),
- (gst_id3_tag_add_src_pad), (gst_id3_tag_init),
- (gst_id3_tag_set_property), (gst_id3_tag_do_caps_nego),
- (gst_id3_tag_src_link), (gst_id3_tag_chain),
- (gst_id3_tag_change_state), (plugin_init):
- deprecate id3tag element and replace with id3demux/id3mux.
- great side effect: this ugly file is now even uglier, yay!
- * ext/mad/gstmad.h:
- remove non-available function
- update for new get_type
+ tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
+ Original commit message from CVS:
+ * tests/check/elements/multifdsink.c: (wait_bytes_served),
+ (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
+ add two more tests, one doing streamheader
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- require mpeg2dec >= 0.4.0
+ gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
+ clean up the bufqueue when shutting down
+ * tests/check/Makefile.am:
+ * tests/check/elements/multifdsink.c: (setup_multifdsink),
+ (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
+ (main):
+ add a test for the leak that was just fixed
-2004-04-17 Benjamin Otte <otte@gnome.org>
+2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_xwindow_id):
- call GST_ELEMENT_ERROR whenever get_xcontext fails. Includes
- assorted cleanup fixes.
+ * gst/tcp/gstmultifdsink.c:
+ DEBUG_FUNCPTR'ing
+ Original commit message from CVS:
+ DEBUG_FUNCPTR'ing
-2004-04-16 David Schleef <ds@schleef.org>
+2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.h: Compile fix for FreeBSD. (bug #140268)
- * sys/xvimage/xvimagesink.h: same
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ whitespace fixes
+ Original commit message from CVS:
+ whitespace fixes
+
+2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps),
+ (gst_adder_query_duration), (gst_adder_query), (forward_event),
+ (gst_adder_src_event), (gst_adder_sink_event),
+ (gst_adder_class_init), (gst_adder_finalize),
+ (gst_adder_request_new_pad), (gst_adder_collected):
+ * gst/adder/gstadder.h:
+ Updated some docs. Added comments and FIXMEs all over the place.
+ Improve debugging info.
+ Fix leak on finalize by not calling the parent.
+ Implement duration query.
+ Make event forwarding threadsafe.
+ Correctly send NEWSEGMENT at start and after flush.
+ Handle EOS correctly.
+ Post error when not negotiated.
+ * tests/check/elements/adder.c: (GST_START_TEST):
+ Added FIXME in the test.
+
+2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ Const-ify GEnumValue and GFlagsValue arrays. Use
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
+ (gst_text_overlay_halign_get_type),
+ (gst_text_overlay_wrap_mode_get_type):
+ * ext/theora/theoradec.c: (theora_handle_type_packet),
+ (theora_handle_data_packet):
+ * ext/theora/theoraenc.c: (gst_border_mode_get_type),
+ (theora_enc_sink_setcaps), (theora_enc_chain):
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_mode_get_type):
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audiostestsrc_wave_get_type):
+ * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
+ * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
+ * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
+ (gst_sync_method_get_type), (gst_unit_type_get_type),
+ (gst_client_status_get_type):
+ * gst/videoscale/gstvideoscale.c:
+ (gst_video_scale_method_get_type):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_pattern_get_type):
+ * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
+ (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
+ (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
+ (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
+ (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
+ (paint_setup_RGB565), (paint_setup_xRGB1555):
+ Const-ify GEnumValue and GFlagsValue arrays. Use
+ GST_ROUND_UP_* macros instead of home-made ones.
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Fix GST_ELEMENT_ERROR with (NULL)
+2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Require core CVS for the new newsegment stuff.
+ Original commit message from CVS:
+ * configure.ac:
+ Require core CVS for the new newsegment stuff.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add div[3456] as fourccs for DivX 3 (fixes #140137).
+2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
+ Register nick for enum value (#341160).
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_caps), (gst_riff_create_audio_caps),
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids):
- * gst-libs/gst/riff/riff-read.h:
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- Add MS RLE support. I added some functions to read out strf chunks
- into strf chunks and the data behind it. This is usually color
- palettes (as in RLE, but also in 8-bit RGB). Also use those during
- caps creation. Lastly, add ADPCM (similar to wavparse - which
- should eventually be rifflib based).
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
- (gst_matroska_demux_init), (gst_matroska_demux_reset):
- * gst/matroska/matroska-demux.h:
- Remove placeholders for some prehistoric tagging system. Didn't add
- support for any tag system really anyway.
- * gst/qtdemux/qtdemux.c:
- Add support for audio/x-m4a (MPEG-4) through spider.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_loop):
- ADPCM support (#135862). Increase max. buffer size because we
- cannot split buffers for ADPCM (screws references) and I've seen
- files with 2048 byte chunks. 4096 seems safe for now.
+2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
+ (plugin_init):
+ backout typefind patch #340375
+ * tests/check/elements/adder.c: (message_received),
+ (GST_START_TEST), (adder_suite):
+ redo, signal-handling of test
- * configure.ac: bump nano to 1
+2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-=== release 0.8.1 ===
+ gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_request_new_pad),
+ (gst_adder_collected):
+ * gst/adder/gstadder.h:
+ Remove bogus segment merging and forwarding, we don't
+ care about timestamps anyway and we just produce a
+ continuous stream.
+ Also create a nice NEWSEGMENT event when we start.
+ Use _scale_int some more.
-2004-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac: releasing 0.8.1, "Comforting Sounds"
+ tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/icles/stress-xoverlay.c:
+ Fix if core was built without parsing support.
-2004-04-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Fix typo in divxversion (3 instead of 4 for "DIVX" fourcc).
- Fixes #140058
+ gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add SEDG (Samsung MPEG-4) fourcc.
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_plugin_init):
- lower rank of dvddemux so that it's not used for mpeg playback.
+ tests/examples/volume/volume.c: Fox if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/examples/volume/volume.c:
+ Fox if core was built without parsing support.
+ * tests/examples/seek/seek.c:
+ Disable the parse_launch example if core was built without parsing
+ support.
-2004-04-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- save libs correctly when checking mad
+ tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Disable the parse_launch example if core was built without parsing
+ support.
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstid3tag.c: (plugin_init):
- lower rank of id3tag as proposed by Benjamin. Fixes #139926.
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/videoscale/gstvideoscale.c:
+ doc reparagraphing and DEBUG_FUNCPTRing
+ Original commit message from CVS:
+ doc reparagraphing and DEBUG_FUNCPTRing
-2004-04-13 David Schleef <ds@schleef.org>
+2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
- * common/m4/gst-feature.m4: Call -config scripts with
- --plugin-libs if it is supported.
- * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect): sequences of
- JPEG images are image/jpeg.
- * gst/debug/Makefile.am:
- * gst/debug/negotiation.c: (gst_negotiation_class_init),
- (gst_negotiation_getcaps), (gst_negotiation_pad_link),
- (gst_negotiation_update_caps), (gst_negotiation_get_property),
- (gst_negotiation_plugin_init): Add a property that acts like
- filter caps.
- * testsuite/gst-lint: Move license checking to be a standard
- test.
+ autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
+ Original commit message from CVS:
+ * autogen.sh: (CONFIGURE_DEF_OPT):
+ libtoolize on Darwin/MacOSX is called glibtoolize
-2004-04-13 David Schleef <ds@schleef.org>
+2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset): Fix memleak.
- patch from Sebastien Cote (bug #139958)
+ tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
+ Disable the adder test, until the build-slaves posses the kindness to
+ either like it or to give valid reason for not doing so
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * examples/gstplay/Makefile.am:
- * examples/gstplay/player.c: (main):
- make the commandline player example use gconf settings
+ tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
+ Original commit message from CVS:
+ * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
+ (adder_suite):
+ Shuffle NULL state change around and raise timeout more
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init),
- (gst_cacasink_sinkconnect), (gst_cacasink_init),
- (gst_cacasink_chain), (gst_cacasink_open), (gst_cacasink_close):
- init/end library during state transition, not object
- creation/disposal. get rid of custom dispose handler.
+ gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
+ (mp4_type_find), (plugin_init):
+ Add typefind to distinguish between "audio/x-m4a" and new type
+ "video/mp4". Fixes #340375
+ * tests/check/elements/adder.c: (adder_suite):
+ Raise timeout to make buildbot happy
+2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-12 Christian Schaller <Uraeus@gnome.org>
+ Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_sink_event),
+ (gst_adder_request_new_pad), (gst_adder_change_state):
+ * gst/adder/gstadder.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
+ (adder_suite), (main):
+ Add sink-event handling to adder. It tries to merge incomming
+ newsegment-events. Added test to check if segment_done is comming
+ through.
- * sys/oss/gstosselement.c: s/lstat/stat/ from freeBSD, since it can
- be a symlink
+2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * ext/theora/theoraparse.c:
+ * ext/vorbis/vorbisparse.c:
+ ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ Original commit message from CVS:
+ 2006-05-05 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoraparse.c (gst_theora_parse_init)
+ (theora_parse_src_convert, theora_parse_src_query):
+ * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
+ query functions on the source pads of the theora and vorbis parse
+ elements. Fixes position querying when doing a remux.
+
+2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/theora/theoraparse.c: Fix flushing.
+ Original commit message from CVS:
+ * ext/theora/theoraparse.c: (parse_granulepos),
+ (theora_parse_drain_queue_prematurely),
+ (theora_parse_queue_buffer), (theora_parse_sink_event):
+ Fix flushing.
+ Fix invalid granulepos outputs when starting with a non-keyframe.
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_data):
- Handle JUNK chunks inside data section. Prevents warnings.
+2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
+ (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
+ Rearrange MPEG system stream detection, fixing some memleaks in the
+ process.
+ Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
+ they clean up their data correctly.
+ Remove unused ogganx caps and move the 'is_annodex' check to inside
+ the 'is_ogg' if statement.
+
+2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (cleanup_decodebin):
+ Properly remove ghostpads. Fixes #340392
+
+2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
+
+ gst/typefind/gsttypefindfunctions.c:
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+
+2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (mpeg_ts_probe_headers), (mpeg_ts_type_find):
+ When typefinding an MP3 in push-based mode, don't penalise the
+ probability down to 74% when we found 5 valid frames just because we
+ can't peek the end of the file.
+ Make the probability for detecting MPEG Transport Streams based on the
+ number of sequential headers we successfully detected.
+
+2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
+ (vorbis_dec_push), (vorbis_dec_chain):
+ Still produce an error when we receive an empty packet.
+
+2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
+ (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
+ Mark buffers with DISCONT after seek and after activating new
+ chains.
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_reset),
+ (theora_get_query_types), (theora_dec_sink_event),
+ (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
+ (theora_dec_change_state):
+ Fix frame counter.
+ Detect and mark DISCONT buffers.
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
+ (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
+ (vorbis_dec_change_state):
+ * ext/vorbis/vorbisdec.h:
+ Use GstSegment.
+ Detect and mark DISCONT buffers.
+ Don't crash on 0 sized buffers.
+
+2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
+ (volume_transform_ip):
+ Increase "volume" property to 10.0. Fixes #340369.
+ Set the process function to NULL when capsnego fails so that
+ we properly error out.
+
+2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (add_sink):
+ * gst/playback/test.c: (main):
+ * gst/playback/test5.c: (dump_element_stats):
+ * gst/playback/test6.c: (main):
+ free cpas using gst_caps_unref, don't leak caps-strings
+
+2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add MS video v1.
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_data):
- Add support for "rec-list" chunks.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ some RTP debug
+ Original commit message from CVS:
+ some RTP debug
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Fix another codecname mismatch.
+ gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
+ (plugin_init):
+ Refine musepack typefinding a bit. Return MAXIMUM
+ probability when we detect stream version 7 to make
+ sure the mpeg audio typefinder doesn't trump us.
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Fix divx caps mismatch and move from video/x-jpeg to image/jpeg
- so that MJPEG plays back.
+ gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Protect against unexpected NULL strf_data buffer.
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
- (gst_mp1videoparse_real_chain), (gst_mp1videoparse_change_state):
- * gst/mpeg1videoparse/gstmp1videoparse.h:
- Fix for some slight mis-cuts in buffer parsing, and for some
- potential overflows or faults-causers. Adds disconts. Also fixes
- #139105 while we're at it.
+ tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST):
+ interpret the out[] buffer in the order the bytes are actually
+ put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
+ Other tests should use BYTE_ORDER since the array is filled in
+ with actual values
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- * sys/v4l2/gstv4l2element.h:
- Workaround for missing struct v4l2_buffer declaration in Suse 9
- and Mandrake 10 linux/videodev2.h header file (#135919).
+ * tests/check/elements/audioconvert.c:
+ dump expected data when audioconvert test fails
+ Original commit message from CVS:
+ dump expected data when audioconvert test fails
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Bail out if no filename was given.
+ tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST):
+ when a test fails, give an indication of which it is
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
- (gst_v4l2_fourcc_from_structure):
- Add Y41B/Y42B YUV formats (see #125732), fix Y41P (was typo'ed to
- Y41B somewhere).
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theoraenc.c:
+ add another include
+ Original commit message from CVS:
+ add another include
-2004-04-09 Benjamin Otte <otte@gnome.org>
+2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gnomevfs/gstgnomevfssink.c:
- (_gst_boolean_allow_overwrite_accumulator),
- (gst_gnomevfssink_class_init):
- fix erase signal - if any handler returns false the file will not be
- overwritten. If no handler is connected, the file will not be
- overwritten either.
- renamed signal to "allow-overwrite"
- * ext/mad/gstid3tag.c: (tag_list_to_id3_tag_foreach):
- free string when adding it to ID3 failed
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- unref event when done
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- free caps
- * gst/typefind/gsttypefindfunctions.c:
- (mpeg_video_stream_type_find):
- fix invalid read
+ * gst/subparse/gstssaparse.c:
+ atoi() needs stdlib.h
+ Original commit message from CVS:
+ atoi() needs stdlib.h
-2004-04-08 David Schleef <ds@schleef.org>
+2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_register): Change rank to PRIMARY.
+ * gst/playback/test4.c:
+ * gst/playback/test5.c:
+ * gst/playback/test6.c:
+ exit needs stdlib.h
+ Original commit message from CVS:
+ exit needs stdlib.h
-2004-04-08 David Schleef <ds@schleef.org>
+2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/colorspace/gstcolorspace.c: Don't advertise a conversion
- we don't support (bug #139532)
+ gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ compile fix; strtol() needs <stdlib.h>
-2004-04-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstmad.c: (gst_mad_handle_event),
- (gst_mad_check_caps_reset), (gst_mad_chain),
- (gst_mad_change_state):
- only set explicit caps if they haven't been set before for
- this stream. MPEG-audio sample rate/channels aren't allowed
- to change in-stream.
- Fixes #139382
+ * ChangeLog:
+ * common:
+ * docs/Makefile.am:
+ * docs/libs/Makefile.am:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/Makefile.am:
+ * docs/upload.mak:
+ use common upload.mak
+ Original commit message from CVS:
+ use common upload.mak
-2004-04-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_base_init),
- (_gst_boolean_did_something_accumulator),
- (gst_gnomevfssink_class_init), (gst_gnomevfssink_dispose),
- (gst_gnomevfssink_init), (gst_gnomevfssink_set_property),
- (gst_gnomevfssink_get_property), (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file), (gst_gnomevfssink_chain),
- (gst_gnomevfssink_change_state):
- Fix erase signal. Don't erase by default. Remove handoff signal.
- Remove erase property. Don't segfault. General cleanup.
+ make GstElementDetails const
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixerelement.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreamselector.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/v4l/gstv4ljpegsrc.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * sys/v4l/gstv4lsrc.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/libs/cddabasesrc.c:
+ make GstElementDetails const
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/gconf/test-gconf.c: (main):
- add missing gst_init
+ gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
+ (gst_adder_init):
+ send events from src-pad to all sink-pads fixes #338657
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
+ (alsasink_parse_spec):
+ query witdh capabilities from alsa, fixes #338919
+
+2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_remove_client_link):
+ * gst/tcp/gstmultifdsink.h:
+ Fix race condition in multifdsink that can lead to spurious
+ duplicate clients. this patch adds a new signal that is fired when
+ multifdsink has removed all references to the fd.
+ Fixes #339574.
+ Updated documentation.
+ API: client-fd-removed signal added
+
+2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
+ When asking g_value_array_new to prealloc elements, we may as well
+ ask for the right number of elements.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
- free the mutexes, too
+2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ patch to make timestamp checking more tollerant to rounding
+ errors given that real discontinuities are to be marked on
+ buffers. Fixes some asf files and #338778.
+ Also avoid some crashers when we receive an event in the
+ NULL state.
+
+2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_get_property),
+ (gst_gnome_vfs_src_send_additional_headers_callback),
+ (gst_gnome_vfs_src_received_headers_callback),
+ (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
+ (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ Remove ICY handling (mostly) from gnomevfssrc, in favour of
+ proper shared support within icydemux.
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
- actually free the URI string
- * ext/mad/gstid3tag.c: (gst_id3_tag_src_event):
- compute offset correctly when passing discont events
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- don't leak discont events
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
- add some missing breaks so caps aren't copied randomly
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream):
- if we realloc memory, we better use it
+ gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
+ (gst_video_rate_swap_prev), (gst_video_rate_chain):
+ fix up docs
+ fix a leak when no caps negotiated
+ fix counting of input frames
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/videorate.c: (assert_videorate_stats),
+ (GST_START_TEST), (videorate_suite):
+ add tests for these
-2004-04-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
+ (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
+ (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (gst_ring_buffer_stop),
+ (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
+ (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
+ (gst_ring_buffer_commit), (gst_ring_buffer_read),
+ (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
+ (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
+ Check arguments passed to public functions instead of
+ crashing.
+
+2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create):
+ GstBaseAudioSrc must be live or it does not work.
+ * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
+ Don't set live to TRUE as this is the default in the parentclass.
+
+2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstmad.c: (normal_seek):
- fix GST_FORMAT_TIME usage
+ * win32/common/config.h:
+ update config.h
+ Original commit message from CVS:
+ update config.h
+
+2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
+ (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
+ Videoscale doesn't pass on pixel-aspect ratio. Handle all
+ fixation cases better. Fixes #338991
+
+2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
+ Handle 0/1 framerate correctly Fixes #331901.
+
+2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (get_float_caps),
+ (GST_START_TEST), (audioconvert_suite):
+ Added check for correct clipping when doing float samples
+ in audioconvert.
+
+2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videorate/gstvideorate.c: Print more debugging info.
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_event),
+ (gst_video_rate_chain):
+ Print more debugging info.
+
+2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
+ (resample_set_state_from_caps):
+ Add support for other formats audioresample can handle such as
+ 32 bits in and float and 64 bits float. Fixes #301759
+
+2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float):
+ correctly clip float samples > 1.0. Fixes #338718
+
+2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_render_text):
+ Don't strip newlines from the text. Also, center lines
+ within multi-line paragraphs (#339405).
-2004-04-05 David Schleef <ds@schleef.org>
+2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/kio/kiosrc.cpp: Undefine KDE_DEPRECATED so we can use
- a deprecated function (hack!)
+ gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
+ Fix wavpack typefinding to work in more cases (don't peek
+ for chunks of multiple hundred kBs at once, but process
+ things step-by-step in smaller units). Fixes #339786.
-2004-04-05 Benjamin Otte <otte@gnome.org>
+2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/esd/esdmon.c: (gst_esdmon_get):
- fix nonterminated vararg and memleak
+ * ChangeLog:
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
-2004-04-05 Benjamin Otte <otte@gnome.org>
+=== release 0.10.6 ===
- * ext/ladspa/gstladspa.c: (gst_ladspa_class_init),
- (gst_ladspa_init), (gst_ladspa_force_src_caps),
- (gst_ladspa_set_property), (gst_ladspa_get_property),
- (gst_ladspa_instantiate), (gst_ladspa_activate),
- (gst_ladspa_deactivate), (gst_ladspa_loop), (gst_ladspa_chain):
- clean up debugging
+2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-05 Stefan Kost <kost@imn.htwk-leipzig.de>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * docs/upload.mak:
+ releasing 0.10.6
+ Original commit message from CVS:
+ releasing 0.10.6
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ladspa/gstladspa.c: (gst_ladspa_class_init):
- check for broken LADSPA parameters (fixes #138635)
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ dist more win32 files
+ Original commit message from CVS:
+ dist more win32 files
-2004-04-05 Benjamin Otte <otte@gnome.org>
+2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps):
- advertise buffer-frames correctly on sinkpads
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
- * ext/mad/gstmad.c: (gst_mad_get_type), (gst_mad_layer_get_type),
- (gst_mad_mode_get_type), (gst_mad_emphasis_get_type),
- (gst_mad_get_event_masks), (gst_mad_get_query_types), (index_seek),
- (normal_seek), (gst_mad_src_event), (gst_mad_handle_event),
- (gst_mad_check_caps_reset), (gst_mad_chain):
- add more debugging, only reset caps when we're not in error state
+ gst/videoscale/gstvideoscale.c: Add call to oil_init().
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: Add call to oil_init().
+ Fixes #338897.
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstmad.c: add debugging category, comment + cleanups
+ * ChangeLog:
+ * configure.ac:
+ * win32/common/config.h:
+ new prerelease
+ Original commit message from CVS:
+ new prerelease
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.net>
+2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- reviewed by Benjamin Otte <otte@gnome.org>
+ ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
+ Original commit message from CVS:
+ 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ patch by: Wim Taymans
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_perform_seek):
+ make sure correct newsegments are sent, so that the decoder
+ and the demuxer agree on timestamps. Fixes playback of a lot
+ of Ogg files that do not start from 0. Fixes #339833.
- * configure.ac:
- fix == in test(1) operator
+2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.net>
+ Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
+ Original commit message from CVS:
+ Patch by: Edward Hervey <edward@fluendo.com>
+ * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
+ * tests/check/Makefile.am:
+ * tests/check/elements/videorate.c: (assert_videorate_stats),
+ (setup_videorate), (cleanup_videorate), (GST_START_TEST),
+ (videorate_suite), (main):
+ Fix an infinite loop if frames are passed in with wrongly ordered
+ timestamps. Fixes #339013.
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
- fix --export-symblos-regex to a working regex.
-
-2004-04-04 Benjamin Otte <otte@gnome.org>
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
- * sys/oss/.cvsignore:
- add for oss_probe
+2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-03 Tim-Phillip Müller <t.i.m@zen.co.uk>
+ gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
+ Original commit message from CVS:
+ Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
+ fix typefinding on some ISO files. Fixes #339212.
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- add missing 'new_media' argument (fixes #138168)
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_seek_event):
- add vararg terminator (fixes #138169)
+ gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
+ Original commit message from CVS:
+ Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ add another H264 fourcc. Fixes #339047.
-2004-04-02 David Schleef <ds@schleef.org>
+2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/gdk_pixbuf/Makefile.am: Make sure gstgdkanimation.h is
- disted (bug #138914)
+ gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
+ Original commit message from CVS:
+ Patch by: Jan Schmidt
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_bufferalloc):
+ Restore old StreamSelector behaviour.
+ Fixes #338419.
-2004-04-01 Benjamin Otte <otte@gnome.org>
+2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_change_state),
- (gst_alsa_close_audio):
- handle case better where a soundcard can't pause
- * ext/ogg/gstoggdemux.c:
- don't crash when we get events but don't have pads yet
-
-2004-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/oss/gstosselement.c: (gst_osselement_probe_caps):
- throw an error if we couldn't probe any caps.
-
-2004-04-01 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dvdnav/gst-dvd:
- Add a really simple sample DVD player
-
-2004-04-01 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_get_type), (gst_a52dec_init),
- (gst_a52dec_push), (gst_a52dec_handle_event),
- (gst_a52dec_update_streaminfo), (gst_a52dec_loop),
- (gst_a52dec_change_state):
- * ext/a52dec/gsta52dec.h:
- Use a debug category, Output timestamps correctly
- Emit tag info, Handle events, tell liba52dec about cpu
- capabilities so it can use MMX etc.
- * ext/dv/gstdvdec.c: (gst_dvdec_loop), (gst_dvdec_change_state):
- Fix a crasher accessing invalid memory
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_init),
- (dvdnavsrc_update_highlight), (dvdnavsrc_loop),
- (dvdnavsrc_get_event_mask), (dvdnav_handle_navigation_event),
- (dvdnavsrc_event), (dvdnavsrc_get_formats), (dvdnavsrc_convert),
- (dvdnavsrc_query):
- Some support for byte-format seeking.
- Small fixes for still frames and menu button overlays
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_get_type),
- (gst_mpeg2dec_alloc_buffer):
- Use a debug category. Adjust the report level of several items to
- LOG. Call mpeg2_custom_fbuf to mark our buffers as 'custom buffers'
- so it doesn't lose the GstBuffer pointer
- * gst/debug/Makefile.am:
- * gst/debug/gstdebug.c: (plugin_init):
- * gst/debug/gstnavseek.c: (gst_navseek_get_type),
- (gst_navseek_base_init), (gst_navseek_class_init),
- (gst_navseek_init), (gst_navseek_seek),
- (gst_navseek_handle_src_event), (gst_navseek_set_property),
- (gst_navseek_get_property), (gst_navseek_chain),
- (gst_navseek_plugin_init):
- * gst/debug/gstnavseek.h:
- Add the navseek debug element for seeking back and forth in a
- video stream using arrow keys.
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_get_type),
- (gst_mpeg2subt_base_init), (gst_mpeg2subt_class_init),
- (gst_mpeg2subt_init), (gst_mpeg2subt_finalize),
- (gst_mpeg2subt_getcaps_video), (gst_mpeg2subt_link_video),
- (gst_mpeg2subt_handle_video), (gst_mpeg2subt_src_event),
- (gst_mpeg2subt_parse_header), (gst_get_nibble),
- (gst_setup_palette), (gst_get_rle_code), (gst_draw_rle_line),
- (gst_merge_uv_data), (gst_mpeg2subt_merge_title),
- (gst_update_still_frame), (gst_mpeg2subt_handle_subtitle),
- (gst_mpeg2subt_handle_dvd_event), (gst_mpeg2subt_loop):
- * gst/mpeg2sub/gstmpeg2subt.h:
- Pretty much a complete rewrite. Now a loopbased element. May still
- require work to properly synchronise subtitle buffers.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private),
- (gst_dvd_demux_send_subbuffer):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_subbuffer):
- Don't attempt to create subbuffers of size 0
- Reduce a couple of error outputs to warnings.
- * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect),
- (gst_y4mencode_chain):
- Output the y4m frame header correctly.
-
-2004-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/adder/gstadder.c: (gst_adder_get_type), (gst_adder_loop):
- throw errors instead of allowing SIGFPE
-
-2004-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/gconf/gconf.c: (gst_gconf_get_string),
- (gst_gconf_render_bin_from_key):
- leak plugging and style fixing
-
-2004-03-31 David Schleef <ds@schleef.org>
-
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_expand_value),
- (gst_audioscale_getcaps): Fix getcaps to expand and union lists.
- (bug #138225)
- * gst/debug/Makefile.am:
- * gst/debug/breakmydata.c: (gst_break_my_data_plugin_init):
- * gst/debug/gstdebug.c: (plugin_init): Merge elements into one
- plugin.
- * gst/debug/negotiation.c: (gst_gst_negotiation_get_type),
- (gst_negotiation_base_init), (gst_negotiation_class_init),
- (gst_negotiation_init), (gst_negotiation_getcaps),
- (gst_negotiation_pad_link), (gst_negotiation_chain),
- (gst_negotiation_set_property), (gst_negotiation_get_property),
- (gst_negotiation_plugin_init): New element to talk about random
- negotiation things happening in a pipeline.
-
-2004-03-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/adder/gstadder.c: (gst_adder_get_type), (gst_adder_loop):
- fix integer addition with help of Stefan Kost
-
-2004-03-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/nl.po: updated Dutch translation (Elros Cyriatan)
-
-2004-03-30 David Schleef <ds@schleef.org>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_alloc_buffer),
- (gst_mpeg2dec_negotiate_format): Handle Y42B-format MPEG
- video, patch from Matthew.Spencer@eu.sony.com (Matthew Spencer)
- (bug #137504)
- * ext/mpeg2dec/gstmpeg2dec.h:
-
-2004-03-30 David Schleef <ds@schleef.org>
-
- * ext/gdk_pixbuf/Makefile.am: Remove spurious rules. (bug #136527)
-
-2004-03-30 David Schleef <ds@schleef.org>
-
- * tools/gst-launch-ext-m.m: Applied patch from gnome@flyn.org (W.
- Michael Petullo) to handle .mov
-
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/oss/gstosselement.c: (gst_osselement_probe_caps),
- (gst_osselement_rate_check_rate):
- probe caps correctly for sound cards that only support one format
-
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/kio/kiosrc.cpp: (process_events):
- update handling event processing if inside KDE - untested
-
-2004-03-29 David Schleef <ds@schleef.org>
-
- * ext/hermes/gsthermescolorspace.c: (plugin_init): decrease rank
- by 2 to not interfere with other colorspaces.
- * ext/pango/gsttextoverlay.c: (plugin_init): change rank to NONE
- * gst/colorspace/gstcolorspace.c: (plugin_init): decrease rank by
- one to not interfere with ffmpeg_colorspace.
-
-2004-03-29 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
- aren't in the caps.
- * gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
- * gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
+ * ChangeLog:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ reverting rtp patches to fix freeze break on -base as explained on the list
+ Original commit message from CVS:
+ reverting rtp patches to fix freeze break on -base as explained on the list
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c:
- fail on error, don't try to set stuff on NULL caps
+ gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
+ Original commit message from CVS:
+ 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ New RTP audio base payloader class. Supports frame or sample based codecs
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
* configure.ac:
- * ext/Makefile.am:
- * ext/kio/Makefile.am:
- * ext/kio/kioreceiver.cpp:
- * ext/kio/kioreceiver.h:
- * ext/kio/kiosrc.cpp:
- * ext/kio/kiosrc.h:
- add experimental kiosrc plugin
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- initialize debugging category only when we're sure registering the
- plugins worked.
-
-2004-03-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * examples/gstplay/player.c: (main):
- * gst-libs/gst/play/play.c: (gst_play_class_init),
- (gst_play_set_location), (gst_play_set_data_src),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization), (gst_play_connect_visualization):
- check return values of element_set_state and return FALSE where
- failed
-
-2004-03-29 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- try harder to check if an event is really a discont
-
-2004-03-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS: adding Azerbaijani (M??tin ??mirov)
+ * po/af.po:
* po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ update libtool versioning
+ Original commit message from CVS:
+ update libtool versioning
-2004-03-28 Benjamin Otte <otte@gnome.org>
-
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (gst_mpeg_demux_parse_packet), (gst_mpeg_demux_parse_pes):
- get rid of non-standard "..." ranges in case statements.
-
-2004-03-27 Martin Soto <martinsoto@users.sourceforge.net>
-
- * gst/mpegstream/gstmpegdemux.c:
- * gst/mpegstream/gstmpegdemux.h: Complete overhaul. All DVD
- specific functionality split to the new dvddemux element.
- * gst/mpegstream/gstdvddemux.c:
- * gst/mpegstream/gstdvddemux.h: New demultiplexer for DVD (VOB)
- streams, derived from mpegdemux.
- * gst/mpegstream/gstmpegparse.c: Discontinuity handling cleaned
- up. SCR based timestamp rewriting can be turned off (will probably
- completely disappear soon).
- * ext/dvdnav/dvdnavsrc.c: Changes resulting from a few months
- hacking. General cleanup. All printf statements replaced by
- debugging messages. Almost complete libdvdnav support.
- (dvdnavsrc_class_init): Got rid of unnecessary signals (replaced
- by events. New properties for audio and subpicture languages.
- (dvdnavsrc_update_highlight): Now uses events.
- (dvdnavsrc_user_op): Cleaned up.
- (dvdnavsrc_get): Renamed to dvdnavsrc_loop (element is now loop
- based). Lots of cleanup, and propper support for most libdvdnav
- events.
- (dvdnavsrc_make_dvd_event): New function.
- (dvdnavsrc_make_dvd_nav_packet_event): New function.
- (dvdnavsrc_make_clut_change_event): New function.
-
-2004-03-26 Benjamin Otte <otte@gnome.org>
-
- * gst/typefind/gsttypefindfunctions.c: (theora_type_find):
- fix bug where typefinding would claim it's theora whenever less then
- 7 bytes of data were available
-
-2004-03-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/law/alaw-decode.c: (alawdec_getcaps), (alawdec_link),
- (gst_alawdec_base_init), (gst_alawdec_class_init),
- (gst_alawdec_init), (gst_alawdec_chain):
- * gst/law/alaw-encode.c: (alawenc_getcaps), (alawenc_link),
- (gst_alawenc_base_init), (gst_alawenc_class_init),
- (gst_alawenc_init), (gst_alawenc_chain):
- * gst/law/mulaw-decode.c: (mulawdec_getcaps), (mulawdec_link),
- (gst_mulawdec_base_init), (gst_mulawdec_class_init),
- (gst_mulawdec_init), (gst_mulawdec_chain):
- * gst/law/mulaw-encode.c: (mulawenc_getcaps), (mulawenc_link),
- (gst_mulawenc_base_init), (gst_mulawenc_class_init),
- (gst_mulawenc_init), (gst_mulawenc_chain):
- Fix capsnego in all four, remove the unused property functions and
- simplify the chain functions slightly. I guess we could use macros
- or something similar for those, since the code is so similar, but
- I'm currently too lazy...
-
-2004-03-24 David Schleef <ds@schleef.org>
-
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms),
- (gst_osselement_close_audio), (gst_osselement_probe_caps),
- (gst_osselement_get_format_structure),
- (gst_osselement_rate_probe_check), (gst_osselement_rate_add_range),
- (gst_osselement_rate_check_rate), (gst_osselement_rate_add_rate),
- (gst_osselement_rate_int_compare): Add code to handle rate probing
- (bug #120883)
- * sys/oss/gstosselement.h: same
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_getcaps):
- Use rate probing provided by osselement.
- * sys/oss/gstosssrc.c: (gst_osssrc_init), (gst_osssrc_getcaps): same
-
-2004-03-24 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_set_property),
- (gst_xvidenc_get_property):
- ulong/int mess-up.
-
-2004-03-24 David Schleef <ds@schleef.org>
-
- * ext/speex/gstspeexdec.c: (gst_speexdec_base_init),
- (gst_speexdec_init):
- * ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
- (gst_speexenc_init): Create the pad template correctly (from
- the static pad template, not a NULL pointer.)
-
-2004-03-25 Benjamin Otte <otte@gnome.org>
-
- * gst/debug/Makefile.am:
- * gst/debug/breakmydata.c:
- add element that quasi-randomly changes bytes in the stream.
- Intended use is robustness checking of demuxers and decoders in
- media tests.
-
-2004-03-24 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio),
- (gst_alsa_probe_hw_params):
- * ext/alsa/gstalsa.h:
- debugging output fixes
+2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-24 Benjamin Otte <otte@gnome.org>
+ * ChangeLog:
+ * configure.ac:
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
+
+2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_set_property):
- don't g_return_if_fail if element is PLAYING, fail silently as every
- other element.
- * gst/effectv/gstquark.c: (gst_quarktv_chain):
- only fix needed for cast lvalue issues in gst-plugins
- * gst/volenv/gstvolenv.c: (gst_volenv_init):
- add proxy_getcaps
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
+ Fix some memory leaks: on finalize, free buffers left in the queue
+ before destroying the queue; in _push(), unref rtp_buf even if
+ the process vfunc returned a NULL buffer as output buffer (#337548);
+ demote some recuring debug messages to LOG level.
-2004-03-24 Benjamin Otte <otte@gnome.org>
+2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/level/gstlevel.c: (gst_level_init):
- add proxying getcaps function, so level doesn't advertise impossible
- caps
+ * gst-plugins-base.spec.in:
+ fix version number macro
+ Original commit message from CVS:
+ fix version number macro
+
+2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: More cleanups.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
+ (gst_ogg_demux_loop):
+ More cleanups.
+ Respect segment stop when emiting EOS or SEGMENT_DONE.
+ Fixes (#337945).
+
+2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gststreamselector.c: Don't leak pad name.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_get_property):
+ Don't leak pad name.
-2004-03-24 David Schleef <ds@schleef.org>
+2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (gst_qtdemux_loop_header), (qtdemux_parse_moov), (qtdemux_parse),
- (qtdemux_node_dump_foreach), (qtdemux_dump_mvhd),
- (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd),
- (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref),
- (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss),
- (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco),
- (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd),
- (qtdemux_parse_tree), (qtdemux_parse_trak): Fix debugging
- messages. Divide the chunk size by the compression ratio
- (needed for MACE audio)
+ * ChangeLog:
+ Mention bug #336617 closed by recent commit
+ Original commit message from CVS:
+ Mention bug #336617 closed by recent commit
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Fix buffer overflow read error.
+ tests/check/: so that FC4 buildslaves can pass.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/gst-plugins-base.supp:
+ Suppress an old libtheora bug (fixed in more recent versions), so
+ that FC4 buildslaves can pass.
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Don't leak events.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
+ (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
+ (gst_ogg_demux_init), (gst_ogg_demux_finalize),
+ (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
+ (gst_ogg_demux_loop):
+ Don't leak events.
+ Remember what error we got when finding chains, if we
+ were shutdown, that would not be an error.
+
+2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_event):
+ Starting the ringbuffer when we did not acquire it can cause
+ a deadlock, is pointless and causes nasty things for
+ subclasses.
+ Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
+
+2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Add some more debugging.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
+ (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
+ (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
+ (gst_ogg_demux_bisect_forward_serialno),
+ (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
+ Add some more debugging.
+
+2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * common:
+ * ext/theora/theoraenc.c:
+ fix width of docs
+ Original commit message from CVS:
+ fix width of docs
- * ext/alsa/gstalsa.h:
- Remove unused entry.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add cinepak.
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
- (gst_videodrop_link), (gst_videodrop_chain):
- Fix, sort of. Was horribly broken with new capsnego. Bah...
+2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-03-23 Jeremy Simon <jesimon@libertysurf.fr>
+ ext/theora/theoradec.c: Some more debug info.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_src_event),
+ (theora_handle_data_packet):
+ Some more debug info.
+ * tests/examples/seek/seek.c: (start_seek), (main):
+ Print element messages too.
- * gst/typefind/gsttypefindfunctions.c: (ape_type_find),
- (plugin_init):
- Add a monkeysaudio typefind function
+2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-03-23 Johan Dahlin <johan@gnome.org>
+ gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
+ Original commit message from CVS:
+ * gst/audioresample/debug.h:
+ replace debug macros with variable number of parameters
+ by a simple alias to gstreamer standard debug macros
+ (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
+ supported by MSVC 6.0 and 7.1)
+ * gst/audioresample/resample.h:
+ define M_PI and rint for WIN32
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstriff.def:
+ * win32/common/libgsttag.def:
+ * win32/common/libgstvideo.def:
+ add new exported functions
+ * win32/vs6:
+ update project files
+
+2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixeroptions.c:
+ (gst_alsa_mixer_options_class_init):
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
+ * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_class_init):
+ * gst-libs/gst/audio/gstaudiosink.c:
+ (gst_audioringbuffer_class_init):
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ (gst_audioringbuffer_class_init):
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
+ * gst-libs/gst/interfaces/colorbalancechannel.c:
+ (gst_color_balance_channel_class_init):
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ (gst_mixer_options_class_init):
+ * gst-libs/gst/interfaces/mixertrack.c:
+ (gst_mixer_track_class_init):
+ * gst-libs/gst/interfaces/tunerchannel.c:
+ (gst_tuner_channel_class_init):
+ * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ (gst_netbuffer_class_init):
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init):
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
+ * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_class_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
+ * sys/v4l/gstv4lcolorbalance.c:
+ (gst_v4l_color_balance_channel_class_init):
+ * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
+ * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
+ * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
+ * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
+ (gst_v4l_tuner_norm_class_init):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
+ * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
+ Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
+
+2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Fix broken GObject macros
+ Original commit message from CVS:
+ * ext/pango/gsttextrender.h:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ * gst-libs/gst/video/gstvideofilter.h:
+ * gst-libs/gst/video/gstvideosink.h:
+ * gst/playback/gstplaybasebin.h:
+ * gst/tcp/gstmultifdsink.h:
+ * sys/v4l/gstv4lelement.h:
+ Fix broken GObject macros
+
+2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
+ More debug to trace why my USB headset is not working with gst
+
+2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_destroy):
+ Clean up our group elements properly in the case where it never
+ got committed - it still got added unconditionally to the bin.
+
+2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Unref unhandled events.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_sink_event),
+ (theora_handle_data_packet), (theora_dec_chain):
+ Unref unhandled events.
+ Protect against empty buffers.
+ Perform QoS on running time.
+
+2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
+ (gst_vorbis_enc_chain):
+ Remove leaks from vorbisenc.
+ Mostly minor changes, the only significant one is that now the
+ buffers we set as 'streamheader' on the caps are copies of the
+ original buffers, to avoid circular refcounting problems.
- * gst-libs/gst/play/play.c (gst_play_audio_fixate)
- (gst_play_video_fixate): Check so the structure has the field
- before trying to fixate them, this makes it possible to have
- fakesinks for video and audio output without printing errors on
- the output console.
+2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-03-22 David Schleef <ds@schleef.org>
+ gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
+ Don't remove our mute-probe if someone else already did so.
+ Don't set a 2nd one if there is already one pending on the pad.
+ * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
+ (do_playbin_seek):
+ When a seek fails, ensure that playbin is still set back to playing.
+ * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
+ (mpeg_ts_type_find), (plugin_init):
+ Add a typefind function for mpeg-ts streams.
- * sys/oss/Makefile.am:
- * sys/oss/oss_probe.c: (main), (probe_check), (add_range),
- (check_rate), (add_rate): Rate probing test app.
+2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
-2004-03-21 Benjamin Otte <otte@gnome.org>
+ * ChangeLog:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/videorate/gstvideorate.c:
+ gst/videorate/gstvideorate.c (gst_video_rate_reset)
+ Original commit message from CVS:
+ 2006-04-06 Andy Wingo <wingo@pobox.com>
+ * gst/videorate/gstvideorate.c (gst_video_rate_reset)
+ (gst_video_rate_init): Caps-related parameters should not be reset
+ by a flush -- move their inits to the instance init function.
+ (gst_video_rate_flush_prev): Don't complain if gst_pad_push
+ is not OK, just return the result.
+ * gst/audiotestsrc/gstaudiotestsrc.c
+ (gst_audio_test_src_class_init)
+ (gst_audio_test_src_get_times): Re-enable is-live=true, as was
+ broken by Stefan's commit on 24 March.
+
+2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
+ Original commit message from CVS:
+ 2006-04-06 Andy Wingo <wingo@pobox.com>
+ * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
+ buffers being pushed out. Fixes oggmux ! multifdsink.
+
+2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
+ (gst_vorbis_dec_init), (vorbis_dec_finalize):
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
+ (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
+ (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
+ (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
+ (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
+ (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
+ (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
+ (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
+ (gst_vorbis_enc_buffer_from_packet),
+ (gst_vorbis_enc_buffer_from_header_packet),
+ (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
+ (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
+ (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
+ (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
+ (gst_vorbis_enc_change_state):
+ * ext/vorbis/vorbisenc.h:
+ Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
+ vorbisenc adhere to the official nomenclature; use boilerplate
+ macro.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
- (_fixate_caps_to_int), (gst_audio_convert_fixate):
- add a fixation function that pretty much does the right thing (fixes
- #137556)
+2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
-2004-03-20 David I. Lehn <dlehn@users.sourceforge.net>
+ gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
+ Original commit message from CVS:
+ 2006-04-04 Andy Wingo <wingo@pobox.com>
+ * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
+ Whoops, fix bug introduced. Bad hacker!
- * configure.ac: GST_PACKAGE default: s/GStreamer/GStreamer Plugins/
+2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
-2004-03-20 Tim-Phillip Müller <t.i.m@zen.co.uk>
+ gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
+ Original commit message from CVS:
+ 2006-04-04 Andy Wingo <wingo@pobox.com>
+ * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
+ Properly handle the case where you get EOS before any buffers are
+ received. Use gst_buffer_make_metadata_writable where appropriate.
- reviewed by: Benjamin Otte <otte@gnome.org>
+2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- terminate gst_event_new_discontinuous correctly (fixes parts of
- #137711)
+ ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
+ Original commit message from CVS:
+ 2006-04-04 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoradec.c (theora_handle_data_packet): This value
+ is often negative -- make it signed so as not to wrap around.
+ Fixes segfaults introduced on 9 March.
-2004-03-19 David Schleef <ds@schleef.org>
+2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/Makefile.am: Enable xoverlay unconditionally,
- since it doesn't depend on X, and it's part of our ABI.
+ ext/theora/: Don't try to store a gdouble in a gboolean.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (theora_dec_src_event):
+ Don't try to store a gdouble in a gboolean.
+ Small cleanups.
-2004-03-19 Iain <iain@prettypeople.org>
+2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
- * gst/interleave/deinterleave.c (deinterleave_sink_link): Use the
- is_int in the structure, not the local variable.
+ ext/ogg/gstoggmux.c: Oggmux sucks.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
+ Oggmux sucks.
+ Make it suck slightly less by writing out the final page.
+ Still can't encode a vorbis-in-ogg file correctly, though.
-2004-03-19 David Schleef <ds@schleef.org>
+2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
- * ext/librfb/gstrfbsrc.c: (gst_rfbsrc_change_state),
- (gst_rfbsrc_init), (gst_rfbsrc_getcaps), (gst_rfbsrc_fixate),
- (gst_rfbsrc_link), (gst_rfbsrc_paint_rect), (gst_rfbsrc_get):
- Improvements in caps negotiation.
+ ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
+ Original commit message from CVS:
+ 2006-04-03 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
+ a g_print.
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
- * po/LINGUAS:
- * po/af.po:
- adding Afrikaans (Petri Jooste)
+ ext/theora/theora.c (plugin_init): Register theoraparse.
+ Original commit message from CVS:
+ 2006-04-03 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theora.c (plugin_init): Register theoraparse.
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c: New files implementing a theora
+ parser. Now we can properly remux ogg/theora+vorbis, yay.
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- throw error instead of g_critical (#137588)
+ ext/vorbis/vorbisparse.c: Add some docs and a copyright.
+ Original commit message from CVS:
+ 2006-04-03 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * Makefile.am:
+ * common:
* configure.ac:
- dist common and m4 correctly
- * po/sv.po:
-
-2004-03-17 David Schleef <ds@schleef.org>
+ don't use AS_LIBTOOL_TAGS, it doesn't work
+ Original commit message from CVS:
+ don't use AS_LIBTOOL_TAGS, it doesn't work
- * pkgconfig/gstreamer-media-info.pc.in: Add Version.
- (bug #137348)
+2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ * ext/pango/gsttextoverlay.c:
+ * sys/v4l/gstv4lsrc.c:
+ remove BT8x8 from description, works for more devices
+ Original commit message from CVS:
+ remove BT8x8 from description, works for more devices
- * po/LINGUAS:
- * po/sv.po:
- adding Swedish translation (Christian Rose)
+2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ Fixed the sample pipeline (see #323798)
- * Makefile.am: use release.mak
+2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ use AS_VERSION and AS_NANO more cleanups
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ * win32/common/config.h.in:
+ use AS_VERSION and AS_NANO
+ more cleanups
- * common/ChangeLog:
- * common/gst-autogen.sh:
- add some explanation about the version detection
- * configure.ac:
- fix X check
+2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
+ uninitialized variable return that would happen.
+
+2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
+ uninitialized variable return that would never happen.
+
+2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ (vorbis_parse_sink_event): Add an event function to flush our
+ state on a seek, and to drain buffers on a premature EOS.
+ (vorbis_parse_push_headers, vorbis_parse_clear_queue)
+ (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
+ (vorbis_parse_chain, vorbis_parse_queue_buffer)
+ (vorbis_parse_drain_queue): Queue up buffers until we can set
+ their timestamps and granulepos values.
+ * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
+ and keep track of data needed for deriving granulepos and
+ timestamps for buffers.
+
+2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ expose pluginsdir so gonlin can use it for tests
+ Original commit message from CVS:
+ expose pluginsdir so gonlin can use it for tests
- * configure.ac: bump nano to 1
+2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-=== release 0.8.0 ===
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ add ccda to libraries
+ Original commit message from CVS:
+ add ccda to libraries
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
- * configure.ac: release 0.8.0, "Pharmaceutical Itch"
+ better/unified long descriptions
+ Original commit message from CVS:
+ Patch by: j^ <j at bootlab dot org>
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_class_init):
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ better/unified long descriptions
+ Fixes #336477
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- update libtool version
- * gst-libs/gst/media-info/Makefile.am:
- actually use libtool version
+ tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
+ (stop_seek):
+ Don't let double and tripple clicks mess up our state.
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: fix speex detection to work with 1.0 but not 1.1
+ gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_video_element),
+ (gen_text_element), (gen_audio_element), (gen_vis_element):
+ Error out gracefully when we can't create any of the usual
+ conversion elements for some reason. Also, don't try to
+ create an audioscale (sic) element that's not used anyway.
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gst-plugins.spec.in:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-gconf-uninstalled.pc.in:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs.pc.in:
- * pkgconfig/gstreamer-media-info-uninstalled.pc.in:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins.pc.in:
- remove @VERSION@ from some of the pc files since core and plugins
- are decoupled.
- created gstreamer-plugins.pc as it's a better name, but keeping
- -libs around for now to get fixes upstream done first.
+ gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Don't post RESOURCE_NOT_FOUND error when we can't find a source
+ element for a particular protocol, that's confusing for users.
+ Instead, post a RESOURCE_FAILED error, so that our own error
+ message is actually shown in totem etc. (#336303).
-2004-03-15 Julien MOUTTE <julien@moutte.net>
+2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * gst-libs/gst/play/play.c: (gst_play_get_framerate),
- (gst_play_get_sink_element): First draft of gst_play_get_framerate.
- * gst-libs/gst/play/play.h:
+ ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_get_icy_metadata):
+ Fix some minor memory leaks (#336194).
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * *.c, *.cc: don't mix tabs and spaces
+ ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfs.c:
+ (gst_gnome_vfs_location_to_uri_string):
+ * ext/gnomevfs/gstgnomevfs.h:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ (gst_gnome_vfs_sink_set_property):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
+ Make gnomevfssink accept filenames as well as URIs for the
+ "location" property, just like gnomevfssrc does (and
+ filesrc/filesink do) (#336190).
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup):
- use the new ffmpegcolorspace
- * gst-plugins.spec.in:
- package new colorspace and media-info
- * configure.ac:
- * pkgconfig/Makefile.am:
- fix some more disting issues
- * pkgconfig/gstreamer-media-info-uninstalled.pc.in:
- * pkgconfig/gstreamer-media-info.pc.in:
- generate media-info pc files
+ tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
+ Original commit message from CVS:
+ * tests/check/generic/clock-selection.c: (GST_START_TEST):
+ set to NULL before unreffing, fixes a valgrind leak.
+ Why was this not triggering the error that an object needs to
+ be NULL before unreffing ?
+ * win32/common/config.h:
+ update
-2004-03-15 Johan Dahlin <johan@gnome.org>
+2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * *.h: Revert indenting
+ gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (convert_encoding),
+ (gst_sub_parse_change_state):
+ * gst/subparse/gstsubparse.h:
+ Text subtitle files may or may not be UTF-8. If it's not, we
+ don't really want to see '?' characters in place of non-ASCII
+ characters like accented characters. So let's assume the input
+ is UTF-8 until we come across text that is clearly not. If it's
+ not UTF-8, we don't really know what it is, so try the following:
+ (a) see whether the GST_SUBTITLE_ENCODING environment variable
+ is set; if not, check (b) if the current locale encoding is
+ non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
+ the current locale encoding is UTF-8 and the environment variable
+ was not set to any particular encoding. Not perfect, but better
+ than nothing (and better than before, I think) (fixes #172848).
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- adding ffmpegcolorspace element
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/common.h:
- * gst/ffmpegcolorspace/dsputil.c: (dsputil_static_init):
- * gst/ffmpegcolorspace/dsputil.h:
- * gst/ffmpegcolorspace/gstffmpeg.c: (plugin_init):
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_pix_fmt_to_caps),
- (gst_ffmpeg_caps_to_pix_fmt):
- * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_caps_remove_format_info),
- (gst_ffmpegcolorspace_getcaps), (gst_ffmpegcolorspace_pad_link),
- (gst_ffmpegcolorspace_get_type), (gst_ffmpegcolorspace_base_init),
- (gst_ffmpegcolorspace_class_init), (gst_ffmpegcolorspace_init),
- (gst_ffmpegcolorspace_chain), (gst_ffmpegcolorspace_change_state),
- (gst_ffmpegcolorspace_set_property),
- (gst_ffmpegcolorspace_get_property),
- (gst_ffmpegcolorspace_register):
- * gst/ffmpegcolorspace/imgconvert.c:
- (avcodec_get_chroma_sub_sample), (avcodec_get_pix_fmt_name),
- (avcodec_get_pix_fmt), (avpicture_fill), (avpicture_layout),
- (avpicture_get_size), (avcodec_get_pix_fmt_loss),
- (avg_bits_per_pixel), (avcodec_find_best_pix_fmt1),
- (avcodec_find_best_pix_fmt), (img_copy_plane), (img_copy),
- (yuv422_to_yuv420p), (yuv422_to_yuv422p), (yuv422p_to_yuv422),
- (C_JPEG_TO_CCIR), (img_convert_init), (img_apply_table),
- (shrink41), (shrink21), (shrink12), (shrink22), (shrink44),
- (grow21_line), (grow41_line), (grow21), (grow22), (grow41),
- (grow44), (conv411), (gif_clut_index), (build_rgb_palette),
- (bitcopy_n), (mono_to_gray), (monowhite_to_gray),
- (monoblack_to_gray), (gray_to_mono), (gray_to_monowhite),
- (gray_to_monoblack), (avpicture_alloc), (avpicture_free),
- (is_yuv_planar), (img_convert), (get_alpha_info_pal8),
- (img_get_alpha_info), (deinterlace_line),
- (deinterlace_line_inplace), (deinterlace_bottom_field),
- (deinterlace_bottom_field_inplace), (avpicture_deinterlace):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- * gst/ffmpegcolorspace/mem.c: (av_malloc), (av_realloc), (av_free):
- * gst/ffmpegcolorspace/mmx.h:
- * gst/ffmpegcolorspace/utils.c: (avcodec_init):
- adding ffmpegcolorspace element supplied by Ronald after cleaning
- up and pulling in the right bits of upstream source.
- I'm sure a better C/compiler wizard could do some cleaning up (for
- example use GLIB's malloc stuff), but as a first pass this
- works very well
+ * docs/plugins/tmpl/.gitignore:
+ * tests/check/libs/.gitignore:
+ * tests/check/pipelines/.gitignore:
+ * tests/examples/volume/.gitignore:
+ ignore more
+ Original commit message from CVS:
+ ignore more
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.h:
- I assume Ronald forgot to commit the change to have cardname
- as a struct member. Expect some public spanking at the next
- opportunity.
+ configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
+ Original commit message from CVS:
+ 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ update core requirement to 0.10.4.1 because of async_playback
+ vmethod on GstBaseSink
-2004-03-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsa.c: (gst_alsa_get_property),
- (gst_alsa_open_audio), (gst_alsa_close_audio):
- * ext/alsa/gstalsa.c:
- Don't open the device if we're a mixer (= padless).
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_class_init),
- (gst_alsa_mixer_init), (gst_alsa_mixer_open),
- (gst_alsa_mixer_close), (gst_alsa_mixer_change_state):
- Open mixer during state change rather than during object
- initialization. Also, get a device name. Currently in a somewhat
- hackish fashion, but I didn't really find something better.
+ use DEBUG_FUNCPTR for collectpads
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
+ * gst/adder/gstadder.c: (gst_adder_init):
+ use DEBUG_FUNCPTR for collectpads
-2004-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * *.c, *.h: run gst-indent
+ * Makefile.am:
+ don't go through check-torture if no check installed
+ Original commit message from CVS:
+ don't go through check-torture if no check installed
+
+2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
+ (gst_gnome_vfs_sink_class_init):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
+ (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_base_init):
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_base_init):
+ * gst/adder/gstadder.c: (gst_adder_get_type):
+ * gst/adder/gstadder.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
+ (gst_audio_test_src_create):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_base_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
+ * gst/volume/gstvolume.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ * tests/old/examples/gob/gst-identity2.gob:
+ Add docs for adder, use GST_ELEMENT_DETAILS macro,
+ define GstElementDetails at the top
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
+ Original commit message from CVS:
+ * win32/common/libgstinterfaces.def:
+ Add a lot of export functions for gst-python
+ * win32/common/libgstinterfaces.dsp:
+ Add a missing include folder in the project configuration
+
+2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create),
+ (gst_base_audio_src_change_state):
+ Fix audio sources, forgot to make the ringbuffer
+ startable...
+
+2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create),
+ (gst_base_audio_src_change_state):
+ unparent instead of unref the ringbuffer.
+
+2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
+ Implement new async_play vmethod to start slaving and allow
+ playback start in case of async PLAY state changes.
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ Enable QoS with new method in base class.
+
+2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
+
+ gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
+ Original commit message from CVS:
+ Patch by: Julien MOUTTE <julien at moutte dot net>
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
+ (gst_video_test_src_do_seek), (gst_video_test_src_create):
+ Partially handle 0 framerate, only EOS after the first frame
+ is missing.
+
+2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
+
+ gst/: Patch for support of YVU9 AVI files (#334822)
+ Original commit message from CVS:
+ Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c:
+ Patch for support of YVU9 AVI files (#334822)
+
+2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
+ Original commit message from CVS:
+ * docs/design/design-decodebin.txt:
+ Added design document for new decodebin
+ (Target Caps): text/x-pango-markup is also a default target caps.
+
+2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ docs/design/design-decodebin.txt: Added design document for new decodebin
+ Original commit message from CVS:
+ * docs/design/design-decodebin.txt:
+ Added design document for new decodebin
+
+2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_dispose):
+ Since we _parent the ringbuffer, we also need to
+ _unparent instead of a plain _unref.
+
+2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add scrub checkbox.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
+ (stop_seek), (scrub_toggle_cb), (main):
+ Add scrub checkbox.
+
+2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
+ Original commit message from CVS:
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
+ (gst_ogg_parse_chain):
+ Fix very inefficient usage of linked lists (#335365).
- * gst/modplug/gstmodplug.cc:
- * gst/modplug/gstmodplug.h:
- set correct timestamps on outgoing buffers
+2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gcc 4.1 unreferenced pointer fixes.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
+ * gst/playback/gstplaybin.c: (handoff):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property):
+ gcc 4.1 unreferenced pointer fixes.
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
+ gst_buffer_ref() now takes a GstBuffer*.
+
+2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
+ Original commit message from CVS:
+ 2006-03-20 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c:
+ (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
+ by Jan Schmidt.
+
+2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
+ (id3v1_type_find), (apetag_type_find), (plugin_init):
+ Can't do tag preferences via probability, as tags would then
+ lose against types that are recognised with MAXIMUM probability
+ (like .wav); so let all tag typefinders return MAXIMUM themselves
+ and order them via the rank. Split ID3v1 and ID3v2 typefinders so
+ that we can prefer APE to ID3v1 (fixes #335028).
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/modplug/gstmodplug.cc:
- handle events - don't do crap when a discont arrives that's not
- necessary
- This allows correct loading and playback of mods in Rhythmbox
+ gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_change_state):
+ * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
+ (gst_ring_buffer_may_start):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Only start playback if we are playing.
+ should fix #330748.
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- * gst-libs/gst/gconf/Makefile.am:
- * pkgconfig/Makefile.am:
- move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
- they get rebuilt properly
- * configure.ac:
- when checking for vorbis, try pkgconfig first.
- * gst/modplug/gstmodplug.cc:
- add fixate function
+ Revert accidental commits to these files.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
+ * win32/common/config.h:
+ Revert accidental commits to these files.
-2004-03-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
- Fix for obvious mistake, where we first shift the offset and then
- read a samplesize element assuming the old offset. Note that this
- part still has something weird, i.e. my movies containing those
- don't actually play well, but at least there's something that looks
- like sound now.
+ tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
+ Original commit message from CVS:
+ Patch by: Michal Benes <michal dot benes at xeris dot cz>
+ * tests/Makefile.am:
+ Don't try to build tests in tests/icles if we
+ don't have X (#323852)
-2004-03-14 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: (speex_type_find),
- (plugin_init):
- Add a typefind function for speex format
+2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Add TXXX frame identifiers for replaygain stuff as used
+ by some taggers (see #323721).
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps),
- (gst_asf_demux_setup_pad):
- Use 25fps as our "fake" fps value (marked for fixage in 0.9.x)
- instead of 0. Reason is simple: some elements have a fps range
- of 1-max instead of 0-max. So now ASF video actually works.
+2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property),
+ (gst_stream_selector_bufferalloc):
+ Preserve the existing buggy streamselector behaviour by performing
+ a fallback buffer allocation when downstream isn't linked yet.
+ This should really be fixed in playbin by blocking pads until it's
+ linked them.
+ Also, use gst_pad_alloc_buffer instead of
+ gst_pad_alloc_buffer_and_set.
- * po/LINGUAS:
- * po/sr.po:
- adding serbian as a language
+2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-13 Benjamin Otte <otte@gnome.org>
+ gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Don't crash on unknown ID3v2 TXXX frames.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- return taglist correctly from _get function, don't gst_pad_push it.
- (fixes #137042)
+2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-03-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
+ ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
+ Chain up to the parent finalize method.
+ Add 32-bit sample size to the template caps.
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add the fourcc that the VMWare codec uses.
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property),
+ (gst_stream_selector_bufferalloc),
+ (gst_stream_selector_request_new_pad):
+ For the active pad, forward buffer-alloc requests, otherwise
+ return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
+ having to memcpy every frame when used by playbin.
+ * gst/tcp/gstmultifdsink.c:
+ (gst_multi_fd_sink_handle_client_write):
+ Get negotiated caps from the sink pad, rather than the sink
+ pad's peer.
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_free_list):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init),
- (gst_alsa_mixer_track_new):
- * ext/alsa/gstalsamixertrack.h:
- Fix ancient leftovers... MixerTrack is a GObject.
+ ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
+ Don't forget to set src->callbacks_pushed to FALSE again when
+ popping them, otherwise re-activation in a different mode won't
+ work (#334620).
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Don't block during probing...
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
+ Original commit message from CVS:
+ Patch by: Sebastien Moutte <sebastien moutte net>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
+ (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
+ (gst_ffmpeg_smpfmt_to_caps):
+ Replace __VA_ARGS__ caps creation macros with varargs functions.
+ Makes things compile on MSVC (#320765), looks nicer, and we can
+ tell the compiler to check for the NULL terminator.
-2004-03-12 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
- * ext/alsa/gstalsa.c: (gst_alsa_get_type), (gst_alsa_class_init),
- (gst_alsa_get_property), (gst_alsa_probe_get_properties),
- (gst_alsa_class_probe_devices), (gst_alsa_class_list_devices),
- (gst_alsa_probe_probe_property), (gst_alsa_probe_needs_probe),
- (gst_alsa_probe_get_values), (gst_alsa_probe_interface_init),
- (gst_alsa_open_audio), (gst_alsa_close_audio):
- * ext/alsa/gstalsa.h:
- Add propertyprobe interface implementation, add some device-name
- property, all this so that it looks good in gnome-volume-control.
-
-2004-03-12 David Schleef <ds@schleef.org>
-
- * configure.ac: the Hermes library controls hermescolorspace, not
- colorspace.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_base_init),
- (gst_mpeg2dec_init): minor pet peeve: disable code with #ifdef,
- not /* */
- * ext/sdl/sdlvideosink.c: Change XID to unsigned long.
- * ext/sdl/sdlvideosink.h: ditto.
- * gst/colorspace/gstcolorspace.c: Fix old comments about Hermes
-
-2004-03-12 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_set_xwindow_id),
- (gst_x_overlay_got_xwindow_id):
- * gst-libs/gst/xoverlay/xoverlay.h:
- replace XID with unsigned long to get rid of the xlibs dependency in
- XOverlay (fixes #137004)
-
-2004-03-13 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/effectv/gstaging.c: (gst_agingtv_base_init),
- (gst_agingtv_setup):
- * gst/effectv/gstdice.c: (gst_dicetv_get_type),
- (gst_dicetv_base_init), (gst_dicetv_class_init),
- (gst_dicetv_setup), (gst_dicetv_init), (gst_dicetv_draw):
- * gst/effectv/gstedge.c: (gst_edgetv_get_type),
- (gst_edgetv_base_init), (gst_edgetv_class_init), (gst_edgetv_init),
- (gst_edgetv_setup), (gst_edgetv_rgb32):
- * gst/effectv/gsteffectv.c:
- * gst/effectv/gstquark.c: (gst_quarktv_link), (gst_quarktv_init),
- (gst_quarktv_set_property):
- * gst/effectv/gstrev.c: (gst_revtv_get_type),
- (gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init),
- (gst_revtv_setup), (gst_revtv_rgb32):
- * gst/effectv/gstshagadelic.c: (gst_shagadelictv_get_type),
- (gst_shagadelictv_base_init), (gst_shagadelictv_class_init),
- (gst_shagadelictv_init), (gst_shagadelictv_setup),
- (gst_shagadelictv_rgb32):
- * gst/effectv/gstvertigo.c: (gst_vertigotv_get_type),
- (gst_vertigotv_base_init), (gst_vertigotv_class_init),
- (gst_vertigotv_setup), (gst_vertigotv_init), (gst_vertigotv_rgb32):
- * gst/effectv/gstwarp.c:
- Port everything that can be ported to videofilter and fix up the caps.
- Can someone with a big-endian machine please check these?
-
-2004-03-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_get_time),
- (gst_osssink_chain), (gst_osssink_change_state):
- Latest fixes for A/V sync, audio playback and such. This is about
- all... MPEG playback issues are mostly related to the async build-
- up of MPEG files, I cannot fix that. Use basicgthread to solve it.
-
-2004-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch from: Stephane Loeuillet
+ gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
+ Original commit message from CVS:
+ Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make sure the buffer we copy into is really always big
+ enough, this time for real (#333488).
- * configure.ac:
- use pkg-config for some libraries, falling back to the old .m4 way
- (fixes #131270)
- * m4/libdv.m4:
- removed
+2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add support for 24bpp DIB (#305279).
- * configure.ac:
- * tools/Makefile.am:
- * tools/Makefile.in:
- * tools/gst-launch-ext-m.m:
- * tools/gst-launch-ext.1.in:
- * tools/gst-visualise-m.m:
- * tools/gst-visualise.1:
- * tools/gst-visualise.1.in:
- reorganizing generation of script tools
+2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-03-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Re-enable QoS after the release.
+ Original commit message from CVS:
+ * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
+ (gst_video_scale_init), (gst_video_scale_src_event):
+ Re-enable QoS after the release.
+ Rework videoscale to use the base class src_event handler.
- * ext/divx/gstdivxdec.c:
- Downgrade priority. We prefer ffdec_mpeg4.
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps), (gst_faad_srcconnect),
- (gst_faad_chain), (gst_faad_change_state):
- Fix capsnego. Doesn't work for some sounds because we don't have
- a 5:1 to stereo element.
- * ext/xvid/gstxvid.c: (plugin_init):
- Add priority.
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- Add discont handling.
+2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Colin Walters <walters@verbum.org>
+ configure.ac: back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ back to CVS.
- * gst/audioconvert/gstaudioconvert.c: Fix typo in width 8
- conversion.
+=== release 0.10.5 ===
-2004-03-09 Benjamin Otte <otte@gnome.org>
+2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init):
- the signals take 2 arguments
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ releasing 0.10.5
+ Original commit message from CVS:
+ releasing 0.10.5
-2004-03-09 David Schleef <ds@schleef.org>
+2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad),
- (gst_alsa_fixate): Add fixate function. (bug #136686)
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_init):
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-03-09 Benjamin Otte <otte@gnome.org>
+2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init), (gst_mikmod_loop),
- (gst_mikmod_change_state):
- * ext/mikmod/gstmikmod.h:
- make mikmod's loop function not loop infinitely and call
- gst_element_yield anymore
- * gst/modplug/gstmodplug.cc:
- fix pad negotiation (fixes #136590)
+ docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ Part of previous cdparanoiasrc docs fixes, forgot to commit.
-2004-03-09 David Schleef <ds@schleef.org>
+2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lcs/Makefile.am: Fix so that the lcs colorspace plugin
- doesn't conflict with the internal colorspace plugin.
- * gst-libs/gst/audio/make_filter: Use `` instead of $() to
- satisfy the crappy-ass shell shipped by a certain vendor.
- * gst/videofilter/make_filter: same (bug #135299)
+ docs/plugins/: Add cdparanoiasrc to docs.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ Add cdparanoiasrc to docs.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ More GstCddaBaseSrc docs.
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: bump nano to 1
+ Add new API to libgsttag: gst_tag_from_id3_user_tag().
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
+ * gst-libs/gst/tag/tag.h:
+ Add new API to libgsttag: gst_tag_from_id3_user_tag().
-=== release 0.7.6 ===
+2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ NULL-terminate array of mpeg4 video file extensions.
+ Fixes crash on PPC (#334226).
- * configure.in: releasing 0.7.6, "There"
+2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ (gst_gnome_vfs_src_check_get_range):
+ gnome_vfs_uri_is_local() alone is not a good indicator
+ whether we can operate in pull-mode with a specific URI,
+ as it returns FALSE for file:// URIs that point to an
+ NFS-mounted path. Be more conservative here: whitelist
+ local files, blacklist http URIs and use the old
+ mechanism for anything else (fixes #334216).
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-play.pc.in:
- synchronize the two
+2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ configure.ac: back to trunk
+ Original commit message from CVS:
+ * configure.ac:
+ back to trunk
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_base_init),
- (cdparanoia_open), (cdparanoia_event):
- fix/add error handling
- * po/POTFILES.in:
- add cdparanoia source
- * tools/Makefile.am:
- make scripts executable
+=== release 0.10.4 ===
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- * ext/vorbis/Makefile.am:
- * sys/Makefile.am:
- remove id3types, vorbisfile and xvideosink from the build (#133783)
-
-2004-03-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- Fix metadata read crash (#136537).
-
-2004-03-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * docs/upload.mak:
+ * win32/common/config.h:
+ releasing 0.10.4
+ Original commit message from CVS:
+ releasing 0.10.4
- * gst-libs/gst/media-info/media-info-priv.c: (gmi_set_mime):
- * gst-libs/gst/media-info/media-info.c: (gst_media_info_read):
- adding mime types, fixing the one-stop function
+2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-03-08 Christian Schaller <Uraeus@gnome.org>
+ gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
+ Original commit message from CVS:
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ Disable max-lateness by setting it to -1 for now, so that
+ we can bed QoS stuff in thoroughly between now and the next
+ release.
- * ext/nas/nassink.c and /ext/nas/nassink.h:
- More NAS love from Arwed von Merkatz
- So lets all sing 'Can you feel the NAS tonight'
-
-2004-03-08 Christian Schaller <Uraeus@gnome.org>
-
- * tools/gst-launch-ext.in:
- Replace vorbisfile with oggdemux/vorbisdec/audioconvert
-
-2004-03-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_base_init),
- (gst_mpeg2dec_init):
- remove the user_data pad for now, because it is being used in
- fixating causing MPEG playback to fixate on 1000 Hz for playback.
- If someone knows how to fix this properly, please do.
-
-2004-03-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * sys/oss/gstosssink.c: (gst_osssink_get_delay),
- (gst_osssink_get_time):
- add a warning, IMO this won't get triggered anymore, remove later
+ gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make sure we don't read beyond the palette buffer in case of
+ broken or manipulated files (#333488, patch by: Fabrizio
+ Gennari)
-2004-03-07 David Schleef <ds@schleef.org>
+2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Added Cinepak
- format (bug #136470)
+ gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Fix for variable not initialized.
-2004-03-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/Makefile.am:
- * gst-libs/gst/media-info/Makefile.am:
- * gst-libs/gst/media-info/media-info-priv.c: (found_tag_callback),
- (error_callback), (gst_media_info_error_create),
- (gst_media_info_error_element), (gmip_init), (gmip_reset),
- (gmi_clear_decoder), (gmip_find_type_pre), (gmip_find_type):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c: (main):
- * gst-libs/gst/media-info/media-info.c: (gst_media_info_init),
- (gst_media_info_class_init), (gst_media_info_instance_init),
- (gst_media_info_set_source), (gst_media_info_read_with_idler),
- (gst_media_info_read_idler), (gst_media_info_read):
- * gst-libs/gst/media-info/media-info.h:
- fixed, should work now
+ * configure.ac:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * win32/common/config.h:
+ prereleasing
+ Original commit message from CVS:
+ prereleasing
+
+2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Small cleanups.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_get_type),
+ (gst_visual_src_setcaps), (gst_vis_src_negotiate),
+ (gst_visual_chain):
+ Small cleanups.
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (gst_theora_dec_reset), (_theora_granule_time),
+ (theora_dec_src_convert), (theora_dec_sink_convert),
+ (theora_dec_src_query), (theora_dec_src_event),
+ (theora_dec_sink_event), (theora_handle_comment_packet),
+ (theora_handle_header_packet), (theora_dec_push),
+ (theora_handle_data_packet), (theora_dec_chain),
+ (theora_dec_change_state):
+ Add simple QoS.
+
+2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
+ (audiocast_register_listener), (gst_gnome_vfs_src_start):
+ Some cleanups.
+
+2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
+ Don't try to activate NULL chains.
+
+2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Fix invalid memory access to region before peek'd data (#332964).
-2004-03-07 Christian Schaller <Uraeus@gnome.org>
+2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
- * ext/nas/nassink.c:
- A bunch of NAS fixes from Arwed von Merkatz
+ closes #333510.
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
+ * ext/pango/gsttextrender.c: (gst_text_render_init):
+ * gst/adder/gstadder.c: (gst_adder_init):
+ Don't leak padtemplates, patch by Christophe Fergeau,
+ closes #333510.
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
- (qtdemux_parse_trak):
- Fix crash (j might be greater than n_samples, in which case we're
- writing outside the allocated space for the array) and memleak.
+ gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
+ Fix invalid memory access: make sure string passed to
+ regexec() is NUL-termianted.
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosssink.c: (gst_osssink_chain):
- And another caller that couldn't handle delay < 0 (unsigned
- integer overflow). Video now continues playing on an audio
- buffer underrun, and the clock continues working. Audio still
- stalls.
+ gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (mp3_type_find):
+ Refactor mpeg/audio typefinding to make it more maintainable
+ and easier to fine-tune. Make probing into middle of the file
+ work properly (fixes #333900, also see #152688).
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosssink.c: (gst_osssink_get_delay),
- (gst_osssink_get_time):
- get_delay() may return values lower than 0. In those cases, we
- should not actually cast to *unsigned* int64, that will break
- stuff horribly. In my case, it screwed up A/V sync in movies
- in totem rather badly.
+ gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ (utf8_type_find_have_valid_utf8_at_offset):
+ Remove part from previous commit that was bogus:
+ g_utf8_validate() does in fact not accept embedded
+ zeroes, so we don't need to check for those (thanks
+ to Mike for the hint).
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
+2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faac/gstfaac.c: (gst_faac_chain):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/libpng/gstpngenc.c: (user_write_data):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
- * gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
- * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
- Fix several misuse of gst_buffer_merge (it doesn't take ownership
- of any buffer), should fix some leaks. I hope I didn't unref buffers
- that shouldn't be...
+ gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ (utf8_type_find_count_embedded_zeroes),
+ (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
+ Make plain/text typefinder more conservative: firstly, check
+ for embedded zeroes, which are perfectly valid UTF-8 characters,
+ but also a fairly good sign that something is not a plain text
+ file; secondly, probe into the middle of the file if possible.
+ If we can't probe into the middle, limit the probability value
+ to be returned to TYPE_FIND_POSSIBLE (see #333900).
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/media-info/media-info-priv.c: (have_type_callback),
- (deep_notify_callback), (tag_flag_score), (found_tag_callback),
- (error_callback), (gmi_reset), (gmi_seek_to_track),
- (gmi_get_decoder), (gmi_set_mime), (gmip_find_type_pre),
- (gmip_find_type_post), (gmip_find_stream_post),
- (gmip_find_track_streaminfo_post):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c: (print_tag),
- (info_print), (main):
- * gst-libs/gst/media-info/media-info.c:
- (gst_media_info_error_create), (gst_media_info_error_element),
- (gst_media_info_instance_init), (gst_media_info_get_property),
- (gst_media_info_new), (gst_media_info_set_source),
- (gst_media_info_read_idler), (gst_media_info_read):
- * gst-libs/gst/media-info/media-info.h:
- first pass at making this work again. This seems to work on
- tagged ogg/vorbis and mp3 files.
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- fix huge leak: gst_buffer_merge doesn't unref the first argument
- itself.
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mad/gstmad.c: (gst_mad_class_init), (gst_mad_update_info):
- report layer/mode/emphasis
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * ext/mad/gstmad.c: (gst_mad_chain): fixed caps leak
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_new):
- signal serial
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (gst_vorbis_dec_init), (vorbis_dec_event):
- add debug category
- make vorbisdec handle _BYTE and _TIME queries
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * ext/mad/gstmad.c: (gst_mad_chain): send the average bitrate read
- from the xing header
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
- (gst_audio_convert_link), (gst_audio_convert_change_state),
- (gst_audio_convert_buffer_from_default_format):
- do conversions from/to float correctly, fix some caps nego errors,
- export correct supported caps in template and getcaps, use correct
- caps in try_set_caps functions
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- For some reason, I only committed a ChangeLog entry yesterday and
- not the corresponding code...
- * ext/mad/gstmad.c: Fix detection of Xing headers
- * gst/tags/gstid3tag.c: Changes to support TLEN tags
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_get_pad_by_pad),
- (gst_ogg_demux_src_query):
- make sure to handle the case where there's no current chain
- gracefully.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * ext/aalib/gstaasink.c: (gst_aasink_fixate), (gst_aasink_init):
- Add fixate function. (bug #131128)
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
- (gst_sdlvideosink_fixate): Add fixate function.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Fix attempt to print a non-pointer using GST_PTR_FORMAT.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt):
- Fix missing break that was causing ulaw to be interpreted as
- raw int.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
- Fix code that ignores return value of gst_buffer_merge().
- (bug #114560)
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_segment):
- * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice): same
- * testsuite/gst-lint: Check for above.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * gst/udp/gstudpsrc.c: (gst_udpsrc_get): Check for unfixed
- caps and throw an element error. (bug #136334)
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
- (gst_faad_chain): Fix negotiation.
- * ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
- key and button events.
- * gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
- dung heap of code.
- * gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
- depends on gconf
- * gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
- function to encourage better negotiation, particularly between
- audioconvert and osssink.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
- more important.
- * gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
- typefinding.
- * gst/vbidec/vbiscreen.c: Add glib header
- * pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
+ gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Make typefind function name for mpeg4 video unique.
-2004-03-06 Christophe Fergeau <teuf@users.sourceforge.net>
+2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Cleanups, post nice errors.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_init),
+ (gst_visual_clear_actors), (gst_visual_dispose),
+ (gst_visual_reset), (gst_visual_src_setcaps),
+ (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
+ (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
+ (gst_visual_chain), (gst_visual_change_state):
+ Cleanups, post nice errors.
+ Handle sink and src events.
+ Implement simple QoS.
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ Use new basesink methods to configure max-lateness.
+ Small doc update.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
+ Debug statement cleanups.
+ * gst/volume/gstvolume.c: (gst_volume_class_init):
+ Simple cleanup.
+
+2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
+ (gst_text_overlay_init), (gst_text_overlay_set_property),
+ (gst_text_overlay_get_property):
+ Revert API/ABI break from March 1. Keep 'halign' and 'valign'
+ as string type properties, but mark them deprecated. Add
+ 'halignment' and 'valignment' properties that use enums
+ instead of strings.
- * ext/mad/gstmad.c: Fix detection of Xing headers
- * gst/tags/gstid3tag.c: Changes to support TLEN tags
+2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Allow palettes with less than 256 colours in AVI files
+ (#333488, patch by: Fabrizio Gennari).
+
+2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
+ Original commit message from CVS:
+ 2006-03-07 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
+ (gst_text_overlay_video_event): Fix wrong EOS handling on text
+ pad. We were releasing the queued text buffer when we should keep
+ it until video pad gets EOS or discard the text buffer because it's
+ too old. That was eating the last subtitle buffer. Add some more
+ debug.
+
+2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
+ (gst_text_overlay_video_chain):
+ Fix invalid memory access (we can't access a buffer after it's been
+ pushed downstream without taking a reference); fix memory leak (if
+ there's no text to render, bail out before allocating stuff).
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query):
- debug updates
+2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
+ ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
+ * ext/pango/gsttextoverlay.h:
+ If input is plain text, escape it before passing it to
+ pango_layout_set_markup().
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- * ext/mad/gstmad.c: (gst_mad_init), (is_xhead),
- (mpg123_parse_xing_header), (gst_mad_chain): parse Xing header in vbr
- files, and report the parsed length as a GST_TAG_DURATION tag.
- * gst/tags/gstid3tag.c: support TLEN (duration) tag
+2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-05 Benjamin Otte <otte@gnome.org>
+ gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
+ Don't ignore flow return from gst_pad_push().
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
- convert channels correctly. convert correctly to unsigned.
+2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
-2004-03-05 Julien MOUTTE <julien@moutte.net>
+ Don't leak references returned by gst_pad_get_parent()
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_getcaps),
+ (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
+ (gst_vorbisenc_convert_sink):
+ * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
+ (gst_audio_duration_from_pad_buffer):
+ * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
+ (gst_audio_filter_chain):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps):
+ * gst-libs/gst/video/video.c: (gst_video_frame_rate),
+ (gst_video_get_size):
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
+ Don't leak references returned by gst_pad_get_parent()
+ (#333663, based on patch by: Christophe Fergeau).
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state): Check if
- we have a window before clearing it.
+2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-03-05 Julien MOUTTE <julien@moutte.net>
+ ext/gnomevfs/gstgnomevfssink.c: change location param details
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
+ change location param details
+ * gst/volume/gstvolume.c: (plugin_init):
+ correct plugin description
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state): Check if we
- have a window before clearing it.
+2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_check_get_range):
+ Override GstBaseSrc::check_get_range() in order to avoid opening
+ the resource just to check whether we can operate in pull-mode or
+ not - we can predict that pretty well from the URI alone. Should
+ fix problems with last.fm (#331690). (Requires latest core CVS).
- * gconf/gstreamer.schemas.in:
- * gst-libs/gst/gconf/Makefile.am:
- version installation path the same way as for 0.6
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- remove comment that was fixed
+2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-03-05 David Schleef <ds@schleef.org>
+ gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
+ Original commit message from CVS:
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
+ (gst_video_sink_class_init):
+ Throw away frames that are later than 20 ms.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_formats),
- (gst_qtdemux_src_convert), (gst_qtdemux_get_src_query_types),
- (gst_qtdemux_get_event_mask), (gst_qtdemux_handle_src_query),
- (gst_qtdemux_handle_src_event), (gst_qtdemux_add_stream):
- Add prototype code for handling seeking and querying.
+2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
-2004-03-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-media.c:
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
- * examples/gstplay/player.c: (main):
- Initialize variables to NULL. Prevents a segfault because the
- (uninitialized) variable is not NULL, resulting in a crash on
- trying to reach error->message.
+2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-05 Benjamin Otte <otte@gnome.org>
+ ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
+ (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
+ put Theora BOS pages before others. This hardcodes
+ the Ogg/Theora I profile, but hey.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format):
- make float=>int conversion work correctly even in cornercases.
-
-2004-03-04 David I. Lehn <dlehn@users.sourceforge.net>
-
- * debian/README.Debian:
- * debian/build-deps:
- * debian/changelog:
- * debian/control:
- * debian/control.in:
- * debian/copyright:
- * debian/gstreamer-a52dec.files:
- * debian/gstreamer-aa.files:
- * debian/gstreamer-alsa.files:
- * debian/gstreamer-alsa.manpages:
- * debian/gstreamer-arts.files:
- * debian/gstreamer-artsd.files:
- * debian/gstreamer-audiofile.files:
- * debian/gstreamer-avifile.files:
- * debian/gstreamer-cdparanoia.files:
- * debian/gstreamer-colorspace.files:
- * debian/gstreamer-doc.files:
- * debian/gstreamer-dv.files:
- * debian/gstreamer-dvd.files:
- * debian/gstreamer-esd.files:
- * debian/gstreamer-festival.files:
- * debian/gstreamer-flac.files:
- * debian/gstreamer-gconf.conffiles:
- * debian/gstreamer-gconf.files:
- * debian/gstreamer-gconf.postinst:
- * debian/gstreamer-gnomevfs.files:
- * debian/gstreamer-gsm.files:
- * debian/gstreamer-http.files:
- * debian/gstreamer-jack.files:
- * debian/gstreamer-jpeg.files:
- * debian/gstreamer-mad.files:
- * debian/gstreamer-mikmod.files:
- * debian/gstreamer-misc.files:
- * debian/gstreamer-mpeg2dec.files:
- * debian/gstreamer-oss.files:
- * debian/gstreamer-plugin-apps.files:
- * debian/gstreamer-plugin-apps.manpages:
- * debian/gstreamer-plugin-libs-dev.files:
- * debian/gstreamer-plugin-libs.files:
- * debian/gstreamer-plugin-template.postinst:
- * debian/gstreamer-plugin-template.postrm:
- * debian/gstreamer-sdl.files:
- * debian/gstreamer-sid.files:
- * debian/gstreamer-vorbis.files:
- * debian/gstreamer-x.files:
- * debian/mk.control:
- * debian/rules:
- Debian package info not maintained here.
-
-2004-03-04 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/aalib/gstaasink.c: (gst_aasink_class_init):
- * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init):
- * ext/divx/gstdivxenc.c: (gst_divxenc_class_init):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init):
- * ext/gsm/gstgsmenc.c: (gst_gsmenc_class_init):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
- * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
- * ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_class_init):
- * gst-libs/gst/colorbalance/colorbalance.c:
- (gst_color_balance_class_init):
- * gst-libs/gst/colorbalance/colorbalancechannel.c:
- (gst_color_balance_channel_class_init):
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init):
- * gst-libs/gst/play/play.c: (gst_play_class_init):
- * gst-libs/gst/propertyprobe/propertyprobe.c:
- (gst_property_probe_iface_init):
- * gst-libs/gst/tuner/tuner.c: (gst_tuner_class_init):
- * gst-libs/gst/tuner/tunerchannel.c:
- (gst_tuner_channel_class_init):
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_base_init):
- * gst/cutter/gstcutter.c: (gst_cutter_class_init):
- * gst/effectv/gstvertigo.c: (gst_vertigotv_class_init):
- * sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init):
- fix signals to use - instead of _
- * ext/libcaca/gstcacasink.h:
- * ext/sdl/sdlvideosink.h:
- fix header rename
-
-2004-03-04 David Schleef <ds@schleef.org>
-
- * testsuite/gst-lint: Add a check for bad signal names.
-
-2004-03-04 Stefan Kost <kost@imn.htwk-leipzig.de>
-
- reviewed by David Schleef
-
- * gst/videofilter/gstgamma.c: (gst_gamma_rgb32): Fix typo that
- modified the alpha channel and caused a warning. (bug #136192)
-
-2004-04-03 Christian Schaller <Uraeus@gnome.org>
-
- * gst-plugins.spec.in:
- Change names of plugins to actually be correct. Try to keep things
- alphabetical to avoid getting beat up by Thomas
-
-2004-03-03 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/gconf/gconf.c: (gst_gconf_get_default_video_sink):
- Using ximagesink as a default if no gconf key found. We should
- probably consider using alsasink instead of osssink for the audio
- part.
-
-2004-03-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- fix --with-plugins, don't think it ever worked before
- * gst-plugins.spec.in:
- even more updates
+ * ext/ogg/gstoggmux.c:
+ changed more than 5 lines
+ Original commit message from CVS:
+ changed more than 5 lines
-2004-03-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/sdl/sdlvideosink.h:
- * sys/ximage/ximagesink.h:
- * sys/xvideo/xvideosink.h:
- * sys/xvimage/xvimagesink.h:
- Fix for move of gstvideosink.h -> videosink.h.
+ ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
+ Original commit message from CVS:
+ ogg muxing of vorbis and theora now has pages ordered correctly again,
+ even with delays.
+ * ext/ogg/README:
+ updated with some examples
+ * ext/theora/theoraenc.c: (granulepos_to_timestamp),
+ (granulepos_add), (theora_buffer_from_packet):
+ * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
+ (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
+ (gst_vorbisenc_chain):
+ implement strategy from ext/ogg/README
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
+ (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
+ (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
+ (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
+ Fix muxer so that oggz-validate is happy with all streams;
+ except for no eos mark, and the BOS page ordering
+ * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
+ (check_buffer_granulepos):
+ * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
+ update tests to check for OFFSET being set as requested
+ fixed type of granulepos, it's not a ClockTime
-2004-03-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
- * gst-libs/gst/xwindowlistener/Makefile.am:
- this is a plugin library, not a library
+ sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
+ Original commit message from CVS:
+ 2006-03-05 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
+ Check that the xvimage we are creating has a correct size before returning it. (#314897)
-2004-03-01 David Schleef <ds@schleef.org>
+2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * AUTHORS: Added some names. Add yourself if you're still
- missing.
+ gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Give id3 and ape tag typefinders a rank slightly higher
+ than PRIMARY to ensure they're always run before any of
+ the other typefinders (in particular wav and mp3) (#324186).
-2004-03-01 David Schleef <ds@schleef.org>
+2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * MAINTAINERS: Add
+ gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add support for '3IVD' fourcc (#333403).
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-plugins.spec.in: clean up spec file
+ configure.ac: Bump requirements to GStreamer CVS for the new error enum.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump requirements to GStreamer CVS for the new error enum.
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
+ Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
+ space left on the device (fixes #333352).
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
- * gst-libs/gst/video/Makefile.am:
- * gst-libs/gst/video/gstvideosink.c:
- * gst-libs/gst/video/gstvideosink.h:
- rename gstvideosink.h to videosink.h to match other headers
- * gst/mixmatrix/Makefile.am:
- fix plugin filename
- * gst/tags/Makefile.am: fix plugin filename
+ win32/vs6: add a project file for libgstvolume update the workspace
+ Original commit message from CVS:
+ * win32/vs6:
+ add a project file for libgstvolume
+ update the workspace
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/tags/Makefile.am: fix plugin filename
+ * ext/ogg/README:
+ * ext/ogg/gstoggmux.c:
+ debug updates
+ Original commit message from CVS:
+ debug updates
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * examples/gstplay/player.c: (got_time_tick), (main):
- add error handler
- display time_tick more readably
- * gst/mixmatrix/Makefile.am:
- fix plugin file name
+ Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
+ Original commit message from CVS:
+ 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/theora/theoraenc.c: (theora_set_header_on_caps):
+ * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
+ (GST_START_TEST):
+ Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
+ Set IN_CAPS on header buffers
-2004-02-29 Christophe Fergeau <teuf@gnome.org>
+2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/plugins/: Add audioresample to docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Add audioresample to docs.
+ * gst/audioconvert/gstaudioconvert.c:
+ Add revision date.
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_base_init), (gst_audioresample_class_init),
+ (gst_audioresample_init), (gst_audioresample_dispose),
+ (audioresample_get_unit_size), (audioresample_transform_caps),
+ (resample_set_state_from_caps), (audioresample_transform_size),
+ (audioresample_set_caps), (audioresample_event),
+ (audioresample_do_output), (audioresample_transform),
+ (audioresample_pushthrough), (gst_audioresample_set_property),
+ (gst_audioresample_get_property), (plugin_init):
+ * gst/audioresample/gstaudioresample.h:
+ Added docs.
+ Small code cleanups.
+
+2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/oss/gstosselement.c: (gst_osselement_probe),
- (device_combination_append), (gst_osselement_class_probe_devices):
- * sys/oss/gstosselement.h:
- Reworked enumeration of oss dsps and mixers so that gst-mixer works
- on my system using alsa oss emulation, fixes bug #135597
+ * gst/videorate/Makefile.am:
+ fix wim's commit
+ Original commit message from CVS:
+ fix wim's commit
-2004-02-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_init),
- (gst_videodrop_chain), (gst_videodrop_change_state):
- * gst/videodrop/gstvideodrop.h:
- Work based on timestamp of input data, not based on the expected
- framerate from the input. The consequence is that this element now
- not only scales framerates, but also functions as a framerate
- corrector or framerate stabilizer/constantizer.
+ * ext/ogg/gstoggmux.c:
+ debug using the actual GstPad, that allows us to see the serialno in the padname
+ Original commit message from CVS:
+ debug using the actual GstPad, that allows us to see the serialno in the padname
+
+2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/plugins/: Added videoscale to docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Added videoscale to docs.
+ * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
+ (gst_video_rate_swap_prev), (gst_video_rate_event),
+ (gst_video_rate_chain):
+ Fix typo in docs.
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
+ (gst_video_scale_init), (gst_video_scale_prepare_size),
+ (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
+ (gst_video_scale_fixate_caps), (gst_video_scale_transform):
+ * gst/videoscale/gstvideoscale.h:
+ Added docs, examples.
+ Some code cleanups.
+ Post errors instead of g_warning.
+
+2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-27 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggmux.c:
+ clean up debug messages
+ Original commit message from CVS:
+ clean up debug messages
- patches from jmmv@menta.net (Julio M. Merino Vidal)
+2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/interleave/deinterleave.c: (deinterleave_chain): Fix
- GST_ELEMENT_ERROR call (bug #135634)
- * gst/interleave/interleave.c: (interleave_buffered_loop),
- (interleave_bytestream_loop): Don't use alloca() (bug #135640)
- * sys/cdrom/gstcdplayer_ioctl_bsd.h: Fix ioctls on NetBSD (bug #135645)
- * sys/oss/gstosssink.c: (gst_osssink_get_delay),
- (gst_osssink_chain): Fix ioctls on NetBSD. (bug #135644)
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_sync_next_frame),
- (gst_v4lmjpegsrc_set_capture), (gst_v4lmjpegsrc_set_capture_m),
- (gst_v4lmjpegsrc_capture_init), (gst_v4lmjpegsrc_requeue_frame):
- Fix GST_ELEMENT_ERROR call.
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_try_palette): Fix
- GST_ELEMENT_ERROR call.
+ * ext/ogg/gstoggmux.c:
+ extra debugging from older version, makes it easier to compare
+ Original commit message from CVS:
+ extra debugging from older version, makes it easier to compare
-2004-02-27 Benjamin Otte <otte@gnome.org>
+2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/audio/audio.h:
- add macro to make sure header isn't included twice
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
- don't use gst_buffer_free
- * gst/playondemand/filter.func:
- don't use gst_data_free. Free data only once.
+ * ext/ogg/gstoggmux.c:
+ some space cleanup and debug fixes
+ Original commit message from CVS:
+ some space cleanup and debug fixes
+
+2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/: Added some more docs to libs and plugins.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Added some more docs to libs and plugins.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Document ringbuffer some more.
+ * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
+ (gst_video_rate_setcaps), (gst_video_rate_reset),
+ (gst_video_rate_init), (gst_video_rate_flush_prev),
+ (gst_video_rate_swap_prev), (gst_video_rate_event),
+ (gst_video_rate_chain), (gst_video_rate_change_state):
+ * gst/videorate/gstvideorate.h:
+ Fix videorate to use segments.
+ Make it work with 0/1 framerates (closes #331903)
+ Handle EOS correctly.
+ Added docs.
+
+2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
+ (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
+ (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
+ In state change function, first chain up to parent class,
+ then handle downwards state change stuff. Remove some
+ commented out cruft from 0.8 code.
-2004-02-26 David Schleef <ds@schleef.org>
+2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
+ (gst_ogm_parse_chain):
+ Don't remove/re-add source pad if the new caps are the same as
+ the old caps anyway (#333042). When removing source pad, don't
+ unref it afterwards - we didn't ref it when adding. Sprinkle some
+ GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
+ after using gst_pad_get_parent(). Return downstream flow return
+ value in chain function.
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- * gst/level/Makefile.am: -marshal.[ch] and -enum.[ch] files
- should not be disted, -marshal.h files should not be installed,
- and -enum.h files _should_ be installed. Fix to make this the
- case.
+2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ docs/plugins/: Fix hierarchy, added some more elements to the docs.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ Fix hierarchy, added some more elements to the docs.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_get_type):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ Fix docs for ffmpegcolorspace.
+
+2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
+ (apetag_type_find), (ape_type_find), (plugin_init):
+ Some typefinding fine-tuning:
+ - rank ID3/APE tags in order of preference via probabilities, so that
+ ID3v2 > APEv2 > APEv1 > ID3v1.
+ - three or four bytes don't really justify MAXIMUM probability,
+ change those to 'very likely' (musepack and monkeysaudio).
+
+2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Added alsa docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsamixer.h:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
+ * ext/alsa/gstalsamixerelement.h:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasink.h:
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
+ (gst_alsasrc_init):
+ * ext/alsa/gstalsasrc.h:
+ Added alsa docs.
+ Small code cleanups.
+
+2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/Makefile.am: Dist new header too,
+ Original commit message from CVS:
+ * ext/theora/Makefile.am:
+ Dist new header too,
+
+2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Fix some more docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/gnomevfs/gstgnomevfssink.h:
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
+ * ext/vorbis/vorbisenc.h:
+ * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
+ (vorbis_parse_chain), (vorbis_parse_change_state):
+ * ext/vorbis/vorbisparse.h:
+ * gst/audioconvert/gstaudioconvert.h:
+ * gst/tcp/gsttcpserversink.h:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ Fix some more docs.
+ Added docs for vorbisdec and vorbisparse.
+ Fix vorbisparse.
+
+2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Updated/added documentation.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/pango/gstclockoverlay.h:
+ * ext/pango/gsttextoverlay.h:
+ * ext/pango/gsttextrender.h:
+ * ext/pango/gsttimeoverlay.h:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * gst/audioconvert/gstaudioconvert.h:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ * gst/tcp/gstmultifdsink.h:
+ Updated/added documentation.
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
+ (gst_text_overlay_halign_get_type),
+ (gst_text_overlay_wrap_mode_get_type),
+ (gst_text_overlay_base_init), (gst_text_overlay_class_init),
+ (gst_text_overlay_init), (gst_text_overlay_set_property),
+ (gst_text_overlay_get_property):
+ Fix up properties to be enums instead of string to make bindings,
+ introspection and automatic GUI creation possible.
+ Add getters for the properties.
+
+2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ added defines of M_PI and M_PI_2
+ * gst/ffmpegcolorspace/avcodec.h:
+ removed #include "stdint.h" for win32 as _stdint.h is
+ autogenerated to win32/common
+ * win32/common/libgstaudio.def:
+ * win32/common/libgsttag.def:
+ added some exports
+ * win32/vs6:
+ some project files bugs corrected
+ * win32/vs7:
+ project files are reset to the default vs7 configuration
+ (they link to msvcr71.dll using default optimizations)
-=== release 0.7.5 ===
+2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
+ Fix some docs.
+
+2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
+ Original commit message from CVS:
+ * ext/alsa/gstalsasrc.c:
+ Set proper class on the ElementDetails:
+ Source/Audio instead of Src/Audio
+
+2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
+ Original commit message from CVS:
+ * gst/videoscale/vs_scanline.c:
+ (vs_scanline_resample_nearest_RGBA):
+ Revert optimization in videoscale. It should go in liboil and have
+ an appropriate liboil function.
+
+2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock):
+ Don't try to provide a clock in the NULL state.
+
+2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
+ (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
+ (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
+ (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
+ Use GstSegment infrastructure to remove duplicated code
+ and handle more seek cases correctly.
+
+2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_transform):
+ Don't ignore return code from ffmpeg convert function.
+ * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
+ Split out some long statements to ease debugging.
+
+2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_init),
+ (gst_vis_src_negotiate), (get_buffer), (plugin_init):
+ Don't use gst_pad_use_fixed_caps, because it prevents downstream from
+ being able to renegotiate the size. Instead, use the negotiation
+ algorithm from the goom plugin to pick an initial output caps.
+ Also, allow theoretical libvisual plugins that might support non-GL
+ output even if they also do GL.
+
+2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
+ Original commit message from CVS:
+ 2006-02-26 Julien MOUTTE <julien@moutte.net>
+ * ext/libvisual/visual.c: (gst_visual_init),
+ (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
+ (plugin_init): Load only non GL plugins. Fix some memleaks and
+ possible negotiation issues.
+
+2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
+
+ gst-libs/gst/tag/tag.h: Adding Annodex tags here.
+ Original commit message from CVS:
+ 2006-02-25 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
+
+2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
+ (cmml_type_find), (plugin_init):
+ Fix CMML type find function to not require a specific minor version
+ of the CMML header.
+ Add an MPEG4 video elementary stream typefind function.
-2004-02-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac: release 0.7.5, "Under The Sea"
+ ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
+ (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
+ (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
+ (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
+ Annodex support in ogg demuxer. Doesn't do very much without the
+ other annodex patches (to come).
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
- (gst_audio_convert_change_state), (gst_audio_convert_get_buffer):
- * gst/videoscale/gstvideoscale.c:
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- assorted debug/warning fixes
+ gst-libs/gst/riff/riff-media.c:
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Pick up palette for MS video v1 (#327028, patch by:
+ Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps),
- (gst_videoscale_init), (gst_videoscale_chain),
- (gst_videoscale_set_property), (plugin_init):
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/videoscale.c: (gst_videoscale_setup),
- (gst_videoscale_scale_rgb), (gst_videoscale_planar411),
- (gst_videoscale_planar400), (gst_videoscale_packed422),
- (gst_videoscale_packed422rev), (gst_videoscale_32bit),
- (gst_videoscale_24bit), (gst_videoscale_16bit),
- (gst_videoscale_bilinear), (gst_videoscale_bicubic),
- (gst_videoscale_scale_plane_slow),
- (gst_videoscale_scale_point_sample),
- (gst_videoscale_scale_nearest),
- (gst_videoscale_scale_nearest_str2),
- (gst_videoscale_scale_nearest_str4),
- (gst_videoscale_scale_nearest_32bit),
- (gst_videoscale_scale_nearest_24bit),
- (gst_videoscale_scale_nearest_16bit):
- add debugging category and use it properly
- fix use of GST_PTR_FORMAT
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_caps_remove_format_info),
+ (gst_ffmpegcsp_get_unit_size):
+ The 'palette_data' field from incoming RGB caps shouldn't be
+ proxied on outgoing YUV caps; also, restrict unit size
+ adjustment in case of paletted data only to the unit that
+ actually has a palette. Fixes #330711.
-2004-02-25 Andy Wingo <wingo@pobox.com>
+2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/interleave/interleave.c (interleave_buffered_loop): Always
- push only when channel->buffer is NULL. Prevents segfaults doing
- the state change after a nonlocal exit, like a scheme exception.
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
+ (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
+ (gst_ffmpegcsp_get_unit_size):
+ Plug some memory leaks.
- * gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
- Handle the case where the intersected caps is empty.
+2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
+ Original commit message from CVS:
+ * sys/ximage/Makefile.am:
+ * sys/xvimage/Makefile.am:
+ Add some _CFLAGS and _LIBS that seem to be missing
+ and/or required for Cygwin (see #317048).
- * gst/law/mulaw-decode.c: (mulawdec_link):
- * gst/law/mulaw.c: (plugin_init):
- fix mulawdec so it actually works again
+2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-24 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * ChangeLog:
+ ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
+ Original commit message from CVS:
+ ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
- reviewed by: David Schleef <ds@schleef.org>
+2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/videofilter/gstgamma.c: (gst_gamma_class_init),
- (gst_gamma_init), (gst_gamma_set_property),
- (gst_gamma_get_property), (gst_gamma_calculate_tables),
- (gst_gamma_rgb24), (gst_gamma_rgb32): Adds gamma correction
- for RGB, with separate r g and b correction factors. (#131167)
+ ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasrc.c:
+ Fix description as pointed out by caugier.
-2004-02-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- only signal tags for bitrate if they're > 0 (#134894)
+ gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
+ Original commit message from CVS:
+ Reviewed by : Edward Hervey <edward@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
+ (qt_type_find):
+ Better 3gp typefinding.
-2004-02-24 David Schleef <ds@schleef.org>
+2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (plugin_init), (gst_qtdemux_loop_header),
- (qtdemux_parse_moov), (qtdemux_parse), (qtdemux_node_dump_foreach),
- (qtdemux_dump_mvhd), (qtdemux_dump_tkhd), (qtdemux_dump_elst),
- (qtdemux_dump_mdhd), (qtdemux_dump_hdlr), (qtdemux_dump_vmhd),
- (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_stts),
- (qtdemux_dump_stss), (qtdemux_dump_stsc), (qtdemux_dump_stsz),
- (qtdemux_dump_stco), (qtdemux_dump_co64), (qtdemux_dump_dcom),
- (qtdemux_dump_cmvd), (qtdemux_parse_tree), (qtdemux_parse_trak):
- Cleanups. Convert g_prints to GST_LOGs. Add qtdemux debug
- category. Attempt to fix timestamp calculation.
+ ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
+ Don't send EOS event here, the base class will send one for us.
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Subpictures without video stream aren't allowed either.
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
+ Fix debug statement copy'n'paste-o.
-2004-02-24 Johan Dahlin <johan@gnome.org>
+2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/gconf/gconf.c: Add \n to g_print error messages
+ ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
+ Fix issues with mixer keeping state when muting/unmuting
+ and when changing the volume whilst muted (see #331763
+ and #331765).
-2004-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gconf/Makefile.am:
- * gconf/gstreamer.schemas:
- * gst-libs/gst/gconf/Makefile.am:
- * gst-libs/gst/gconf/gconf.c:
- version gconf schemas and install locations
-
-2004-02-23 Benjamin Otte <otte@gnome.org>
-
- * ext/xine/xineinput.c: (gst_xine_input_dispose):
- (gst_xine_input_subclass_init):
- call parent dispose.
- change pad template for CD reader correctly
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c: (plugin_init):
- * ext/xine/xineaudiosink.c:
- wrap audio sinks, too
- * gst-libs/gst/resample/private.h:
- * gst-libs/gst/resample/resample.c: (gst_resample_init),
- (gst_resample_reinit), (gst_resample_scale),
- (gst_resample_nearest_s16), (gst_resample_bilinear_s16),
- (gst_resample_sinc_slow_s16), (gst_resample_sinc_s16),
- (gst_resample_sinc_ft_s16), (gst_resample_nearest_float),
- (gst_resample_bilinear_float), (gst_resample_sinc_slow_float),
- (gst_resample_sinc_float), (gst_resample_sinc_ft_float):
- * gst-libs/gst/resample/resample.h:
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
- (gst_audioscale_class_init), (gst_audioscale_link),
- (gst_audioscale_get_buffer), (gst_audioscale_init),
- (gst_audioscale_chain), (gst_audioscale_set_property),
- (gst_audioscale_get_property):
- * gst/audioscale/gstaudioscale.h:
- s/resample_*/gst_resample_*/i to not clobber namespaces
-
-2004-02-23 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps),
- (gst_riff_create_iavs_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_audio_caps), (gst_asf_demux_add_audio_stream),
- (gst_asf_demux_video_caps), (gst_asf_demux_add_video_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
- (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
- (gst_matroska_demux_plugin_init): First batch implementing audio and
- video codec tags in demuxers.
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c: (plugin_init):
- * ext/xine/xineinput.c:
- add input plugin wrapper. Playback from files, http, mms and cdda
- works.
- * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
- remove leftover G_GNUC_UNUSED
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_stream),
- (gst_asf_demux_identify_guid):
- improve debugging output
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- reported by: Padraig O'Briain <padraig.obriain@sun.com>
+ gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
+ (parse_subrip), (gst_sub_parse_format_autodetect):
+ Set right caps given that we send escaped text. Also,
+ honour <i></i>, <b></b> and <u></u> markers that can be found
+ in .srt files (fixes #310202).
- * autogen.sh:
- replace test -e with test -x for mkinstalldirs to be more portable.
- (fixes #134816)
+2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-22 Benjamin Otte <otte@gnome.org>
+ gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/mixerutils.c:
+ (element_factory_rank_compare_func):
+ Make order in which elements are tried more determinable.
- reported by: Stefan Kost <kost@imn.htwk-leipzig.de>
+2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
+ (remove_element_chain), (cleanup_decodebin),
+ (gst_decode_bin_change_state): Make decodebin reusable by
+ fixing remove_element_chain first and then introduce a
+ cleaner in state change to ->NULL. (Closes #331678)
+ ------------------------------------------------------
+
+2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
+ use 0666 mask when creating files so umask gets applied
+ correctly. Fixes #331295.
+
+2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
+ Original commit message from CVS:
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
+ (gst_ssa_parse_dispose), (gst_ssa_parse_init),
+ (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
+ (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
+ (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
+ (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
+ * gst/subparse/gstssaparse.h:
+ * gst/subparse/gstsubparse.c: (plugin_init):
+ Add very basic parser for SSA subtitle streams (as often
+ found in matroska files).
- * gst/audioconvert/gstaudioconvert.c: (plugin_init):
- set rank to PRIMARY
- * gst/volume/gstvolume.c: (plugin_init):
- set rank to NONE
- fixes #134960
+2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
+ gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (mimetype_is_raw):
+ That should be text/x-pango-markup, not text/x-pango-layout.
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/gsttextoverlay.c: Polishing.
+ Original commit message from CVS:
+ 2006-02-19 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
+ Polishing.
+
+2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/gsttextoverlay.c: Fix state change deadlock.
+ Original commit message from CVS:
+ 2006-02-19 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
+ (gst_text_overlay_finalize), (gst_text_overlay_init),
+ (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
+ (gst_text_overlay_render_text),
+ (gst_text_overlay_text_pad_link),
+ (gst_text_overlay_text_event), (gst_text_overlay_video_event),
+ (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
+ (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
+ Fix state change deadlock.
+
+2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
+ Original commit message from CVS:
+ 2006-02-19 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
+ (gst_text_overlay_finalize), (gst_text_overlay_init),
+ (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
+ (gst_text_overlay_render_text),
+ (gst_text_overlay_text_pad_link),
+ (gst_text_overlay_text_event), (gst_text_overlay_video_event),
+ (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
+ (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
+ * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
+ and subtitles files.
+
+2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gstdecodebin.c: pango layout should be considered as row.
+ Original commit message from CVS:
+ 2006-02-19 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
+ should be considered as row.
+
+2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gststreaminfo.*: Introduce language informations.
+ Original commit message from CVS:
+ 2006-02-19 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
+ (cb_probe):
+ * gst/playback/gststreaminfo.h: Introduce language informations.
+
+2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
+ (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
+ Set shared memory segments to be deleted as soon as we have attached,
+ that way they get cleaned up automatically if we crash.
+
+2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/: Those functions are called with lock held.
+ Original commit message from CVS:
+ 2006-02-18 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
+ * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
+ functions are called with lock held.
+
+2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
+ * ChangeLog:
+ Forgot Changelog.
+ Original commit message from CVS:
+ Forgot Changelog.
+
+2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
+
+ ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
+ Original commit message from CVS:
+ 2006-02-18 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
+ (gst_text_overlay_finalize), (gst_text_overlay_init),
+ (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
+ (gst_text_overlay_render_text),
+ (gst_text_overlay_text_pad_link),
+ (gst_text_overlay_text_pad_unlink),
+ (gst_text_overlay_text_event),
+ (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
+ (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
+ (gst_text_overlay_change_state): Refactoring of textoverlay
+ without collectpads. This now supports sparse subtitles coming
+ from a demuxer instead of a sub file. Seeking is still broken
+ though. Need to discuss with wtay some more on how to handle
+ seeking correctly.
+ * ext/pango/gsttextoverlay.h:
+ * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
+ subtitles coming from the demuxer.
+
+2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisenc.c: Use some more scaling functions.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
+ (gst_vorbisenc_convert_sink):
+ Use some more scaling functions.
+
+2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
+ (gst_cd_paranoia_paranoia_callback),
+ (gst_cd_paranoia_src_signal_is_being_watched),
+ (gst_cd_paranoia_src_read_sector):
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ Add back 'transport-error' and 'uncorrected-error' signals and
+ make them actually be fired when bad stuff happens (#319340).
-2004-02-22 Benjamin Otte <otte@gnome.org>
+2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
+ (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
+ (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
+ (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
+ (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
+ (gst_ring_buffer_pause), (gst_ring_buffer_stop),
+ (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
+ (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
+ (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
+ (gst_ring_buffer_clear):
+ Small cleanups.
+ Added some G_LIKELY.
+
+2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/TODO: Update TODO
+ Original commit message from CVS:
+ * gst-libs/gst/audio/TODO:
+ Update TODO
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_offset):
+ When trying to play samples ASAP and we don't have a
+ previous sample, try to play at position 0 instead of
+ an invalid position.
+
+2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open),
+ (gst_alsasink_reset):
+ Also release lock when we get an error in _reset();
+ fix an error message.
+
+2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
+ (gst_alsasink_init), (get_channel_free_structure),
+ (caps_add_channel_configuration), (gst_alsasink_getcaps),
+ (gst_alsasink_close):
+ * ext/alsa/gstalsasink.h:
+ Add support for more than 2 channels (#326720).
- * configure.ac:
- export [_]*{gst,Gst,GST}.* symbols from plugins
+2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-22 Christophe Fergeau <teuf@gnome.org>
+ gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
+ with 4 or 6 channels, assume a default channel layout to make things
+ work (not sure there's anything else we can do in those cases).
- reviewed by: Benjamin Otte <otte@gnome.org>
+2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lame/gstlame.c: (add_one_tag):
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value),
- (gst_vorbisenc_metadata_set1):
- * gst/tags/gstid3tag.c:
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add):
- apply fixes from bugs #135042 (lame can't write tags) and #133817
- (add GST_ALBUM_VOLUME_{COUNT,NUMBER} tags)
+ gst-libs/gst/audio/multichannel.c: Minor docs fix.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ Minor docs fix.
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
+ Add support for WAVEFORMATEX, eg. PCM audio with more than two
+ channels and a channel layout map.
-2004-02-22 Ramon Garcia <ramon_garcia_f@yahoo.com>
+2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac: Export only gst_plugin_desc from plugins.
- Note that this change only makes any effect with Linux using libtool
- 1.5.2 or higher. Otherwise it is silently ignored, but it would build
- fine. And don't try to have several versions of libtool in different
- directories.
+ gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
+ Original commit message from CVS:
+ Reviewed by Edward Hervey <edward@fluendo.com>
+ * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
+ C-level optimization of the RGBA nearest neighbour function.
+ Eventually this might end up in liboil with vectorized versions.
-2004-02-20 Andy Wingo <wingo@pobox.com>
+2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
- interleave respectively.
+ gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ (gst_audio_get_channel_positions):
+ When we have more than 2 channels, but no channel layout is
+ specified in the caps, return some default channel layout
+ to the caller and warn about about a possibly buggy element
+ (could be buggy filtercaps as well of course) (#317038).
- * gst/interleave/deinterleave.c: New plugin: deinterleave
- (replaces on oneton).
- * gst/interleave/interleave.c: New plugin: interleave.
- * gst/interleave/plugin.h: Support file.
- * gst/interleave/plugin.c: Support file.
+2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: Remove intfloat and oneton, add interleave.
+ pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
+ Original commit message from CVS:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ Add gst-libs/gst/cdda to list of lib search paths.
- * ext/sndfile/gstsf.c: Handle events better.
+2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
- * gst/audioconvert/gstaudioconvert.c: Change to support int2float
- and float2int operation. int2float has scheduling problems as
- noted in in2float_chain.
+ ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
+ Original commit message from CVS:
+ 2006-02-15 Andy Wingo <wingo@pobox.com>
+ * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
+ timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
+ to the Lord Jesus that I do not have to touch the ogg muxer ever
+ again.
-2004-02-20 Benjamin Otte <otte@gnome.org>
+2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c:
- * ext/xine/xineaudiodec.c:
- * ext/xine/xinecaps.c:
- add first version of xine plugin wrapper. Currently only wraps the
- QDM2 win32 DLL, and even that only in proof-of-concept quality.
- * configure.ac:
- * ext/Makefile.am:
- add xine plugin wrapper, disabled by default. Use --enable-xine to
- build. Note that it'll segfault on gst-register if you don't remove
- the goom and tvtime post plugins from xine.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (qtdemux_parse), (qtdemux_parse_trak), (qtdemux_audio_caps):
- add extradata parsing for QDM2.
- change around debugging prints.
+ gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
+ quicktime movie files can also contain 'uuid' atoms.
-2004-02-19 Benjamin Otte <otte@gnome.org>
+2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lame/gstlame.c: (gst_lame_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- use gst_tag_list_insert when you want to insert tags
+ gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
+ Original commit message from CVS:
+ * gst/audioconvert/plugin.c: (plugin_init):
+ Register the GstAudioChannelPosition enum type with the type
+ system in the plugin_init function, so that it is known before
+ any element actually makes use of multi-channel stuff. This is
+ required for example if one wants to be able to deserialise/use
+ a caps string with channel positions before any pipeline has
+ been setup and started, like with gst-launch.
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac: Move massink to gst-rotten
- * ext/Makefile.am:
- * ext/mas/Makefile.am:
- * ext/mas/massink.c:
- * ext/mas/massink.h:
+ gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
+ (gst_ring_buffer_samples_done), (wait_segment),
+ (gst_ring_buffer_commit), (gst_ring_buffer_clear):
+ Add some compiler G_(UN_)LIKELY help.
+ SIGNAL the ringbuffer waiters when going to PAUSED as well to
+ make sure they can exit their functions. Should fix #330748
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init): Disable gdk_pixbuf
- typefinding, since it seems to be worse than nothing.
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find): Add ftyp
- atom to recognize .mp4 and .m4a files as video/quicktime.
+ Windows does not have long long; copy the generated _stdint.h
+ Original commit message from CVS:
+ * Makefile.am:
+ * configure.ac:
+ * win32/MANIFEST:
+ * win32/common/_stdint.h:
+ Windows does not have long long; copy the generated _stdint.h
+ * win32/common/interfaces-enumtypes.c:
+ (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
+ (gst_mixer_track_flags_get_type),
+ (gst_tuner_channel_flags_get_type):
+ * win32/common/multichannel-enumtypes.c:
+ (gst_audio_channel_position_get_type):
+ update
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/sine/demo-dparams.c: (quit_live),
- (dynparm_log_value_changed), (dynparm_value_changed), (main):
- Use double dparams, not float.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init): Change sync default to FALSE, since multiple
- sync'd elements don't really work correctly.
- * gst/volume/gstvolume.c: (volume_class_init), (volume_init),
- (volume_update_volume), (volume_get_property): Change dparam
- to double.
+ gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Always sync on first sample we receive when starting.
-2004-02-18 Julien MOUTTE <julien@moutte.net>
+2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/ximage/ximagesink.c:
- (gst_ximagesink_xwindow_update_geometry),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_handle_xevents),
- (gst_ximagesink_change_state), (gst_ximagesink_expose),
- (gst_ximagesink_init): Rework the way software video scaling works. So
- now we check on each chain call if the video frames are feeling the
- window. If not we try to renegotiate caps. On failure we memorize that
- and we won't try again for that PLAYING sessions.
- * sys/ximage/ximagesink.h: Adding a boolean to store the caps renego
- failure.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_init): initialize the
- synchronous flag.
-
-2004-02-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup):
- break up _link so we can give a better debug message for errors
-
-2004-02-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
- set up debug category
-
-2004-02-18 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_handle_xevents), (gst_ximagesink_expose): Reorganizing
- the way renegotiation work. The event handling function is not taking
- care of external windows and renegotiate method check for pad flags
- NEGOTIATING. Should fix : #133209
-
-2004-02-17 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_expose): Checking if the
- pad is negotiating before trying renegotiation.
-
-2004-02-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_type_find):
- pass on all possible mime types as typefind hints
-
-2004-02-17 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Fix a
- possible SHM leak if we crash. All other apps using XShm are doing
- that.
-
-2004-02-17 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_expose): Renegotiate size on expose.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_expose): Update window
- size on expose.
-
-2004-02-16 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * testsuite/alsa/sinesrc.c:
- cosmetic fix to fix compile issue with gcc 2.95.4
-
-2004-02-16 Julien MOUTTE <julien@moutte.net>
-
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio),
- (gst_alsa_timestamp_to_bytes): Alsa should trigger an error if it
- failed opening the audio device.
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
- (gst_ximagesink_xwindow_new), (gst_ximagesink_xwindow_destroy),
- (gst_ximagesink_xwindow_resize), (gst_ximagesink_xwindow_clear),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_handle_xevents),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_change_state), (gst_ximagesink_chain),
- (gst_ximagesink_set_xwindow_id): Clearing window in READY TO PAUSED.
- Removing some useless g_return_if_fail like wingo suggested.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xvimage_destroy), (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_xwindow_destroy),
- (gst_xvimagesink_xwindow_resize), (gst_xvimagesink_xwindow_clear),
- (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_get_fourcc_from_caps),
- (gst_xvimagesink_change_state), (gst_xvimagesink_chain),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_colorbalance_list_channels),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value): Clearing window in READY TO
- PAUSED. Removing some useless g_return_if_fail like wingo suggested.
-
-2004-02-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
- throw error when not negotiated instead of asserting
-
-2004-02-15 Julien MOUTTE <julien@moutte.net>
-
- * gst/switch/gstswitch.c: (gst_switch_loop): More fixes for
- correct data refcounting.
-
-2004-02-15 Julien MOUTTE <julien@moutte.net>
-
- * gst/switch/gstswitch.c: (gst_switch_change_state),
- (gst_switch_class_init): Cleaning the sinkpads correctly on state
- change, mostly the EOS flag.
-
-2004-02-15 Julien MOUTTE <julien@moutte.net>
-
- * examples/gstplay/player.c: (got_eos), (main): Adding some
- output for debugging.
- * gst-libs/gst/play/play.c: (gst_play_state_change): Stop our
- timeouts if we go to any state different from PLAYING.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek): Fix some
- more EOS bugs in riff lib.
+ gst/playback/gstplaybin.c: Update vis bin docs.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_vis_element):
+ Update vis bin docs.
+ Move queue after tee so we don't queue video buffers but
+ audio samples instead. Fixes problems where the video queue
+ is filled and the audio queue empty.
-2004-02-14 Julien MOUTTE <julien@moutte.net>
+2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/play/play.c: (gst_play_connect_visualization): Disable
- visualization until i find a way to fix switch correctly.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head): Fix a bug when
- EOS arrives.
- * gst/switch/gstswitch.c: (gst_switch_release_pad),
- (gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
- (gst_switch_loop), (gst_switch_dispose), (gst_switch_class_init):
- Reworked switch to get a more correct behaviour with events and refing
- of data stored in sinkpads.
- * gst/switch/gstswitch.h: Adding an eos flag for every sinkpad so that
- we don't pull from a pad in EOS.
-
-2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- remove v1 tag even if we can't read it (makes sure we don't detect
- it again)
-
-2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_pcm_wait),
- (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsa.h:
- try xrun recovery when wait failed. Make xrun recovery function
- return TRUE/FALSE to indicate success. (might fix #134354)
+ gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ No need to push an EOS event here, GstBaseSrc will do that for us
+ when we return FLOW_UNEXPECTED.
-2004-02-13 David Schleef <ds@schleef.org>
+2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Use scale functions when possible.
+ Fix error messages.
+ Free clockid when after waiting for EOS.
+ Use G_(UN_)LIKLY when it makes sense.
+ Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
+
+2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Remove stray semi-colon (fixes #330888).
+
+2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/: Fix up the XShm call testing so that we catch errors, and don't cause new ones by attempting to detach from a s...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
+ Fix up the XShm call testing so that we catch errors, and don't
+ cause new ones by attempting to detach from a segment we failed
+ to attach to. Fixes #312439.
+
+2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Added flv file typefind (video/x-flv).
+
+2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
+ Also added the caps to the default set of riff video caps.
+
+2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
+ Original commit message from CVS:
+ 2006-02-09 Andy Wingo <wingo@pobox.com>
+ * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
+ time and the end time of the last packet in the page.
+ (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
+ on the pages in our queue, set the duration as well. Reflow a
+ debug statement.
+ (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
+ Fixes bad muxing order.
+
+2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_setcaps), (gst_basertppayload_push):
+ update seqnum before setting it on the packet; this makes sure
+ that the timestamp and seqnum properties match after pushing
+ a buffer
- * gst/sine/demo-dparams.c: (dynparm_log_value_changed),
- (dynparm_value_changed), (main): Convert from float to double.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_init): same.
+2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-13 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ changelog foo
+ Original commit message from CVS:
+ changelog foo
- * gst/silence/gstsilence.c: (gst_silence_class_init),
- (gst_silence_set_clock), (gst_silence_get),
- (gst_silence_set_property), (gst_silence_get_property):
- * gst/silence/gstsilence.h: Add sync property.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init), (gst_sinesrc_set_clock), (gst_sinesrc_get),
- (gst_sinesrc_set_property), (gst_sinesrc_get_property):
- * gst/sine/gstsinesrc.h: Add sync property.
+2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-13 David Schleef <ds@schleef.org>
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * win32/common/config.h:
+ kapowpowpow
+ Original commit message from CVS:
+ kapowpowpow
+
+2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst-libs/gst/audio/gstringbuffer.c
+ Original commit message from CVS:
+ 2006-02-09 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstringbuffer.c
+ (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
+ overflow after 13.5 hours of recording. Kapow!
+ * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
+ the buffer size -- we don't care about underrun/overrun reporting
+ right now, just need to return a useful value.
+
+2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * gst/intfloat/gstint2float.c: (conv_f32_s16),
- (gst_int2float_chain_gint16): Change stdint usage to glib types.
+=== release 0.10.3 ===
-2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- * ext/Makefile.am:
- * gst-libs/ext/Makefile.am:
- move ffmpeg plugin to gst-ffmpeg module
-
-2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ Releasing 0.10.3
+ Original commit message from CVS:
+ Releasing 0.10.3
- * configure.ac: use GST_ARCH to detect architecture
+2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-12 Julien MOUTTE <julien@moutte.net>
+ configure.ac: Drat. Bump libtool version number for new API.
+ Original commit message from CVS:
+ * configure.ac:
+ Drat. Bump libtool version number for new API.
+ Prelease 0.10.2.3 (of 0.10.3)
- * gst/vbidec/vbiscreen.c: Fixing thomasvs fixes. Missing header.
+2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
+ 0.10.2.2 prerelease (of 0.10.3).
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.2.2 prerelease (of 0.10.3).
- * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
- classify LADSPA plugins based on number of src/sink pads
- (#133663, Stefan Kost)
- * gst/sine/gstsinesrc.c: (gst_sinesrc_init):
- fix dparams registration
- (#133528, Stefan Kost)
- * gst/vbidec/vbiscreen.c: (vbiscreen_set_current_cell):
- fix use of isprint and use g_ascii_isprint instead
- (#133316, Stefan Kost)
+2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
+ Original commit message from CVS:
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
+ Revert Andy's newsegment change pending a more correct
+ fix.
- Convert a few inner loops to use liboil. This is currently
- optional, and is only enabled if liboil is present (duh!).
- * configure.ac: Check for liboil-0.1
- * gst/intfloat/Makefile.am:
- * gst/intfloat/gstint2float.c: (conv_f32_s16), (scalarmult_f32),
- (gst_int2float_chain_gint16):
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_class_init),
- (tablelookup_u8), (gst_videobalance_planar411):
- * gst/videotestsrc/Makefile.am:
- * gst/videotestsrc/gstvideotestsrc.c: (plugin_init):
- * gst/videotestsrc/videotestsrc.c: (splat_u8), (paint_hline_YUY2),
- (paint_hline_IYU2), (paint_hline_str4), (paint_hline_str3),
- (paint_hline_RGB565), (paint_hline_xRGB1555):
+2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
- * ext/lcs/gstcolorspace.c: (colorspace_find_lcs_format),
- (gst_colorspace_caps_get_fourcc), (colorspace_setup_converter),
- (gst_colorspace_getcaps), (gst_colorspace_link),
- (gst_colorspace_base_init), (gst_colorspace_init),
- (gst_colorspace_chain), (gst_colorspace_change_state),
- (plugin_init): Merge Ronald's patch (bug #117897) and update
- for new caps and negotiation. Seems to work, although it
- shows off bugs in lcs.
+2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-11 David Schleef <ds@schleef.org>
+ * gst/tcp/gstmultifdsink.c:
+ doc fixes
+ Original commit message from CVS:
+ doc fixes
- * ext/alsa/Makefile.am: Fix linking against libgstinterfaces.
- (bug #133886) Noticed by bugs@leroutier.net (Stephane LOEUILLET)
+2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
+ Original commit message from CVS:
+ :
+ * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
+ (qt_type_find), (plugin_init):
+ detect more files as 3gp
+ group and reorder the iso file formats
- * ext/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init),
- (gst_rfbsrc_change_state), (gst_rfbsrc_init),
- (gst_rfbsrc_set_property), (gst_rfbsrc_get_property):
- Add server and port properties
+2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
+ Original commit message from CVS:
+ * ext/vorbis/vorbis.c: (plugin_init):
+ Register musicbrainz tags, so apps don't have to.
- * m4/a52.m4:
- * m4/aalib.m4:
- * m4/as-ffmpeg.m4:
- * m4/as-liblame.m4:
- * m4/as-slurp-ffmpeg.m4:
- * m4/check-libheader.m4:
- * m4/esd.m4:
- * m4/freetype2.m4:
- * m4/gconf-2.m4:
- * m4/glib.m4:
- * m4/gst-alsa.m4:
- * m4/gst-artsc.m4:
- * m4/gst-ivorbis.m4:
- * m4/gst-matroska.m4:
- * m4/gst-sdl.m4:
- * m4/gst-shout2.m4:
- * m4/gst-sid.m4:
- * m4/gtk.m4:
- * m4/libdv.m4:
- * m4/libfame.m4:
- * m4/libmikmod.m4:
- * m4/ogg.m4:
- * m4/vorbis.m4:
- fix underquotedness of macros (#133800)
- * m4/as-avifile.m4:
- * m4/xmms.m4:
- removed because no longer used
+2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
+ (gst_tag_to_vorbis_tag):
+ Make sure we called gst_tag_register_musicbrainz_tags()
+ before possibly mapping a vorbiscomment string from/to a
+ musicbrainz tag.
- * configure.ac:
- require gettext 0.11.5 so ulonglong.m4 gets checked out and copied
- by autopoint (fixes #132996)
+2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
+ In case we can't find the required number of consecutive
+ mpeg audio frames to positively identify an MPEG audio
+ stream, check if there's at least a valid mpeg audio
+ frame right at offset 0 and if so suggest mpeg/audio
+ caps with a very low probability (#153004).
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_base_init):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_base_init):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_base_init):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_base_init):
- fix memleaks
+2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
+ Original commit message from CVS:
+ 2006-02-07 Andy Wingo <wingo@pobox.com>
+ * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
+ a TIME segment if we get timestamped buffers. Requires recent
+ fixes in core to work properly.
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_chain): Fix logic bug causing spurious errors.
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_base_init),
- (gst_jpegdec_init), (gst_jpegdec_chain): Fix negotiation.
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_base_init),
- (gst_jpegenc_class_init), (gst_jpegenc_init),
- (gst_jpegenc_getcaps), (gst_jpegenc_link), (gst_jpegenc_resync),
- (gst_jpegenc_chain), (gst_jpegenc_set_property),
- (gst_jpegenc_get_property): Fix negotiation. Add some properties.
- * ext/jpeg/gstjpegenc.h: Fix negotiation.
+2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-10 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Don't print the URI as part of the error message, it
+ makes error dialogs look rather ugly, especially if
+ the URI is very long or has characters in it that
+ need escaping.
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init),
- (gst_mikmod_srcfixate), (gst_mikmod_srclink), (gst_mikmod_loop):
- * ext/mikmod/gstmikmod.h:
- fix caps negotiation in mikmod
- * ext/ogg/gstoggdemux.c: (gst_ogg_print):
- output debug information
+2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-08 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Error out if we have only text or subtitles, but nothing
+ else. Also error out if we have subtitles but no video
+ stream.
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/navigation/Makefile.am:
- * gst-libs/gst/xoverlay/Makefile.am:
- remove unused GST_OPT_CFLAGS from Makefiles
- include X_CFLAGS and X_LIBS in xoverlay. (#131948)
+2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-07 David Schleef <ds@schleef.org>
+ ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
+ Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
+ Post an error message on the bus when we encounter an
+ error, which will hopefully be more meaningful than the
+ 'Internal Flow Error' message users get to see if we
+ just return GST_FLOW_ERROR.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_handle_event): Don't
- push events to pads that haven't been created (#133508)
+2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-07 Jan Schmidt <thaytan@mad.scientist.com>
+ configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
+ Original commit message from CVS:
+ 2006-02-07 Andy Wingo <wingo@pobox.com>
+ * configure.ac (GST_MAJORMINOR): Update core version req to
+ 0.10.2.2, for the collectpads API addition (#330244).
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_src_convert),
- (gst_dvdec_sink_convert), (gst_dvdec_handle_sink_event),
- (gst_dvdec_video_getcaps), (gst_dvdec_video_link),
- (gst_dvdec_loop), (gst_dvdec_change_state):
- Second attempt at committing a working dvdec element.
+2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-06 David Schleef <ds@schleef.org>
+ ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ Return FALSE from plugin_init() when GnomeVFS can't
+ be initialised for some reason (#328423).
- Build fixes for OS X: (see #129600)
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strh),
- (gst_riff_read_strf_vids), (gst_riff_read_strf_auds),
- (gst_riff_read_strf_iavs):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml):
- * gst/playondemand/Makefile.am:
- * gst/rtp/rtp-packet.c:
+2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
-2004-02-05 David Schleef <ds@schleef.org>
+ ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
+ Original commit message from CVS:
+ 2006-02-06 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
+ Stick to seeking theory until i find the bug.
+ * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_loop): Revert
- last change, because it Just Doesn't Compile.
+2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ Make theoraenc and the tests leak free. Like, really.
+ Original commit message from CVS:
+ * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
+ (theora_enc_finalize), (theora_enc_sink_setcaps),
+ (theora_set_header_on_caps), (theora_enc_chain),
+ (theora_enc_change_state):
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ Make theoraenc and the tests leak free. Like, really.
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- skip undecodable id3v2 tag instead of keeping it
+2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 David Schleef <ds@schleef.org>
+ Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
+ Original commit message from CVS:
+ (theora_enc_finalize), (theora_enc_sink_setcaps):
+ Add a finalize method to ensure we clean up state even if
+ someone omitted the state change back to NULL.
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
+ (gst_vorbisenc_chain):
+ Free some more leaked bits.
+ * tests/check/pipelines/theoraenc.c: (start_pipeline),
+ (stop_pipeline):
+ Wait for state changes to happen if they're ASYNC.
+ This ought to teach those fancy pants buildbots a lesson.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Unref leaked buffer. (Noticed by Ronald)
+2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 David I. Lehn <dlehn@users.sourceforge.net>
+ gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Add mapping for ID3 International Standard Recording Code
+ tag "TSRC"
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- Sync requires with other checks. >= vs =.
+2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+ ext/vorbis/vorbisenc.c: Don't leak tag names.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
+ Don't leak tag names.
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_loop):
- * ext/dv/gstdvdec.h:
- rework the caps negotiation so that dvdec works again instead
- of just segfaulting.
+2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.7.4 ===
+ Split libgsttag docs into multiple sections.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tags.c:
+ Split libgsttag docs into multiple sections.
-2004-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS: GStreamer Plugins 0.7.4 "For Great Justice" released
- * configure.ac: changed for release
+ Add libgsttag to the docs.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ Add libgsttag to the docs.
-2004-02-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
- * gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- reworked patch by David Lehn to fix libdir and includedir for
- uninstalled libraries
- removed play and gconf from gstreamer-libs since they have their
- own pkgconfig files
+ ext/pango/gsttextoverlay.c: Fix clockoverlay.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
+ (gst_text_overlay_init), (gst_text_overlay_src_event),
+ (gst_text_overlay_collected): Fix clockoverlay.
-2004-02-04 David Schleef <ds@schleef.org>
+2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt): Fix a caps
- memleak.
+ docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
+ Original commit message from CVS:
+ * docs/libs/compiling.sgml:
+ Fix typo: it's pkg-config, not pkg-gconfig
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/tmpl/gstgconf.sgml:
+ There is no libgstgconf in 0.10, remove it
+ from the docs.
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
+
+ docs/libs/tmpl/gstcolorbalance.sgml: Updated.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_src_event), (gst_text_overlay_collected):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_class_init), (gst_sub_parse_init),
+ (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
+ (parse_mpsub), (parser_state_init), (handle_buffer),
+ (gst_sub_parse_chain), (gst_sub_parse_sink_event),
+ (plugin_init):
+ * gst/subparse/gstsubparse.h: Introduce seeking code.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- use correct GST_TAG_ENCODER tag
+2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ Add comment about LANGUAGE tag inconsistency (we want
+ ISO-639-1, but extract three-letter identifiers?)
+ * po/POTFILES.in:
+ Add two translatable files.
- * ext/alsa/gstalsa.c: (gst_alsa_change_state):
- be sure to stop the clock when going to paused
- * sys/oss/gstosssink.c: (gst_osssink_change_state):
- reset number of transmitted when going to ready.
- fixes #132935
+2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-05 Charles Schmidt <cschmidt2@emich.edu>
+ gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ (gst_tag_register_musicbrainz_tags_internal),
+ (gst_tag_register_musicbrainz_tags):
+ Forward-port some tags stuff from the 0.8 branch. This is
+ mostly the addition of musicbrainz tags and their mapping
+ to vorbistags, and a vorbistag mapping of the language tag.
- reviewed by Benjamin Otte
+2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- extract track count (fixes #133410)
+ gst/playback/gstplaybin.c: Fix broken code refactoring.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
+ refactoring.
-2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_caps_nego):
- that should be !=, not == (fixes #132519)
+ Add Dirac typefinding and add dirac format to oggmux.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ Add Dirac typefinding and add dirac format to oggmux.
-2004-02-04 David Schleef <ds@schleef.org>
+2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
- Make sure set_explicit_caps() is called before adding pad.
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
+ * configure.ac:
+ Improve error message for liboil missingness.
+ Original commit message from CVS:
+ Improve error message for liboil missingness.
-2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- bump nano to 2, first prerelease
- put back AM_PROG_LIBTOOL
-
-2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * testsuite/alsa/Makefile.am:
- these are user test apps, not automatic testsuite tests
-
-2004-02-04 David Schleef <ds@schleef.org>
-
- Convert GST_DEBUG_CAPS() to GST_DEBUG():
- * gst/mpeg1videoparse/gstmp1videoparse.c:
- (mp1videoparse_parse_seq):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- * sys/xvideo/gstxwindow.c: (_gst_xwindow_new):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_sinkconnect),
- (gst_xvideosink_getcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- * testsuite/gst-lint: more tests
-
-2004-02-04 David Schleef <ds@schleef.org>
-
- Replace use of GST_PAD_FORMATS_FUNCTION() and similar macros
- with the code that they would expand to.
- * ext/flac/gstflacdec.c: (gst_flacdec_get_src_formats),
- (gst_flacdec_get_src_query_types),
- (gst_flacdec_get_src_event_masks):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_formats),
- (gst_gnomevfssrc_get_query_types),
- (gst_gnomevfssrc_get_event_mask):
-
-2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_dispose):
- fix memleak by properly disposing sinesrc
-
-2004-02-04 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_expose):
- * gst-libs/gst/xoverlay/xoverlay.h: Adding the _expose method to tell
- an overlay to redraw the image because it has been exposed.
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_ximage_put), (gst_ximagesink_expose),
- (gst_ximagesink_xoverlay_init), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h: Implement expose method from XOverlay
- interface
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_expose),
- (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h: Implement expose method from XOverlay
- interface
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_type_find):
- more memleak fixage
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
- * gst/typefind/gsttypefindfunctions.c:
- fix memleaks shown by gst-typefind
-
-2004-02-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * common/glib-gen.mak:
- add hack rule to touch .Plo files
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- remove glib_root variable
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
- set explicit caps before adding the element, so the autopluggers can
- plug correctly.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (mpeg2_sys_type_find), (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (dv_type_find):
- fix memleaks in typefind functions. gst_type_find_suggest takes a const
- argument.
-
-2004-02-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/colorbalance/colorbalance-marshal.list:
- * gst-libs/gst/colorbalance/colorbalance.c:
- * gst-libs/gst/colorbalance/colorbalance.h:
- * gst-libs/gst/colorbalance/colorbalancemarshal.list:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/mixer/mixer-marshal.list:
- * gst-libs/gst/mixer/mixer.c:
- * gst-libs/gst/mixer/mixer.h:
- * gst-libs/gst/mixer/mixermarshal.list:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/tuner/Makefile.am:
- * gst-libs/gst/tuner/tuner-marshal.list:
- * gst-libs/gst/tuner/tuner.c:
- * gst-libs/gst/tuner/tuner.h:
- * gst-libs/gst/tuner/tunermarshal.list:
- use new glib-gen.mak snippet to clean up Makefile.am
- fix various bugs in Makefile.am's
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
- handle chain parsing correctly in the multichain case
- * ext/theora/theoradec.c: (gst_theora_dec_init), (_theora_ilog),
- (theora_dec_from_granulepos), (theora_dec_to_granulepos),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- handle events and queries correctly
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * .cvsignore:
- Ignore generated file _stdint.h.
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/colorbalance/colorbalance.h:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/mixer/mixer.h:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/tuner/Makefile.am:
- * gst-libs/gst/tuner/tuner.h:
- Generate enum type code with glib-mkenums.
- * gst-libs/gst/colorbalance/.cvsignore:
- * gst-libs/gst/mixer/.cvsignore:
- * gst-libs/gst/play/.cvsignore:
- * gst-libs/gst/tuner/.cvsignore:
- Ignore generated files.
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * gst-libs/gst/audio/.cvsignore:
- Ignore generated file.
- * gst-libs/gst/audio/Makefile.am:
- Do not install example filter.
+ gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1):
+ Don't put essential function call into
+ g_return_*() macro, otherwise it'll all be
+ replaced by NOOPs when compiling with
+ G_DISABLE_CHECKS defined.
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
+2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * examples/switch/.cvsignore:
- Ignore generated file.
+ * ChangeLog:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggparse.c:
+ * gst/tcp/gsttcpserversink.c:
+ * sys/v4l/v4lsrc_calls.c:
+ * sys/v4l/v4lsrc_calls.h:
+ Just make it compile with --disable-gst-debug.
+ Original commit message from CVS:
+ Just make it compile with --disable-gst-debug.
+
+2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
+ (gst_alsasink_class_init), (gst_alsasink_init),
+ (gst_alsasink_write), (gst_alsasink_reset):
+ * ext/alsa/gstalsasink.h:
+ Add lock to protect alsa calls.
+ Implement reset to flush samples ASAP, does not work
+ with dmix though.
+
+2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock):
+ Ugh.. getting late I guess...
+
+2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock),
+ (gst_base_audio_sink_set_property),
+ (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
+ Don't try to provide a clock when we are not negotiated since
+ we might not be able to make it run.
+
+2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1):
+ Unlinking two source pads is ... hard.
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * common/m4/ax_create_stdint_h.m4:
- * configure.ac:
- add AX_CREATE_STDINT_H to get correct type definitions for a52dec in
- _stdint.h.
- * Makefile.am:
- remove generated _stdint.h in DISTCLEANFILES
- * ext/a52dec/gsta52dec.c:
- include _stdint.h for a52dec. (should fix #133064)
-
-2004-02-02 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add),
- (gst_tag_to_vorbis_comments):
- Add replaygain support to vorbistag
-
-2004-02-02 Jeremy Simon <jesimon@libertysurf.fr>
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
- (gst_ffmpeg_caps_to_extradata):
- Fix SVQ3 caps flag properties
- Use glib macro for bytes swap
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/audiofile/gstafsink.c: (gst_afsink_plugin_init):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/sndfile/gstsf.c: (plugin_init):
- * gst/avi/gstavi.c: (plugin_init):
- * sys/dxr3/dxr3init.c: (plugin_init):
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l2/gstv4l2.c: (plugin_init):
- remove textdomain calls
- * po/nl.po:
- update Dutch translation
-
-2004-02-02 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_set_audio_sink): Moving volume in the audio thread for
- instantaneous volume change. Maybe i will add another volume in front
- of visualization later, not sure yet though.
-
-2004-02-02 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_handle_xevents): Better X events handling, only take
- the latest events for configure and motion.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents): same.
-
-2004-02-02 Jon Trowbridge <trow@gnu.org>
-
- reviewed by: David Schleef <ds@schleef.org>
-
- Fix memory leaks:
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_register):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_register):
-
-2004-02-02 David Schleef <ds@schleef.org>
-
- code cleanup. Change bzero() to memset(). Remove duplicate ; at ends
- of lines.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_event):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/xvid/gstxviddec.c: (gst_xviddec_src_link):
- * gst-libs/gst/play/play.c: (gst_play_get_sink_element):
- * gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
- * gst/effectv/gstedge.c: (gst_edgetv_sinkconnect):
- * gst/effectv/gstvertigo.c: (gst_vertigotv_sinkconnect):
- * gst/intfloat/float22int.c: (gst_float2_2_int_getcaps),
- (gst_float2_2_int_link):
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
- * gst/rtjpeg/RTjpeg.c: (RTjpeg_init_mcompress):
- * gst/tcp/gsttcpsink.c: (gst_tcpsink_init_send):
- * gst/tcp/gsttcpsrc.c: (gst_tcpsrc_init_receive):
- * gst/udp/gstudpsink.c: (gst_udpsink_init_send):
- * gst/udp/gstudpsrc.c: (gst_udpsrc_init_receive):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_init):
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
- * testsuite/gst-lint: Add tests for bzero and ;;
-
-2004-02-02 David Schleef <ds@schleef.org>
-
- * gst/debug/efence.c: Add fallback if MAP_ANONYMOUS isn't defined.
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/aalib/gstaasink.c: (gst_aasink_open):
- * ext/alsa/gstalsa.c: (gst_alsa_link), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
- (gst_afsink_close_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
- (gst_afsrc_close_file):
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_user_op), (dvdnavsrc_get):
- * ext/esd/esdmon.c: (gst_esdmon_get):
- * ext/esd/esdsink.c: (gst_esdsink_chain), (gst_esdsink_open_audio):
- * ext/faac/gstfaac.c: (gst_faac_chain):
- * ext/faad/gstfaad.c: (gst_faad_chain):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_error_callback),
- (gst_flacdec_loop):
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (gst_gnomevfssrc_open_file):
- * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
- * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_chain):
- * ext/lcs/gstcolorspace.c: (gst_colorspace_srcconnect_func):
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_do_typefind), (gst_id3_tag_chain):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain),
- (gst_ogg_demux_push):
- * ext/raw1394/gstdv1394src.c:
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_lock),
- (gst_sdlvideosink_initsdl), (gst_sdlvideosink_create):
- * ext/sndfile/gstsf.c: (gst_sf_open_file), (gst_sf_close_file),
- (gst_sf_loop):
- * ext/speex/gstspeexenc.c: (gst_speexenc_chain):
- * ext/swfdec/gstswfdec.c: (gst_swfdec_loop):
- * ext/tarkin/gsttarkindec.c: (gst_tarkindec_chain):
- * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_loop):
- * ext/xvid/gstxviddec.c: (gst_xviddec_setup), (gst_xviddec_chain):
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_setup), (gst_xvidenc_chain):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_element_data), (gst_riff_read_seek),
- (gst_riff_peek_list), (gst_riff_read_list), (gst_riff_read_header):
- * gst/adder/gstadder.c: (gst_adder_loop):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_segment),
- (gst_asf_demux_process_stream), (gst_asf_demux_get_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_init),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- * gst/avi/gstavimux.c: (gst_avimux_stop_file):
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c: (gst_goom_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/intfloat/float22int.c: (gst_float2_2_int_chain):
- * gst/intfloat/gstfloat2int.c: (gst_float2int_loop):
- * gst/intfloat/gstint2float.c: (gst_int2float_chain_gint16):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_uint), (gst_ebml_read_sint),
- (gst_ebml_read_float), (gst_ebml_read_header):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
- (gst_matroska_demux_parse_blockgroup):
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- * gst/oneton/gstoneton.c: (gst_oneton_chain):
- * gst/silence/gstsilence.c: (gst_silence_get):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/smpte/gstsmpte.c: (gst_smpte_loop):
- * gst/speed/gstspeed.c: (speed_loop):
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * gst/volenv/gstvolenv.c: (gst_volenv_chain):
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_loop):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_open),
- (dxr3audiosink_set_mode_pcm), (dxr3audiosink_set_mode_ac3),
- (dxr3audiosink_close):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_open), (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_open),
- (dxr3videosink_close), (dxr3videosink_write_data):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/oss/gstosssink.c: (gst_osssink_chain):
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_buffer_free):
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay),
- (gst_v4l_set_window), (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_get_capabilities), (gst_v4l_open),
- (gst_v4l_set_chan_norm), (gst_v4l_get_signal),
- (gst_v4l_get_frequency), (gst_v4l_set_frequency),
- (gst_v4l_get_picture), (gst_v4l_set_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_capture_init),
- (gst_v4lsrc_requeue_frame), (gst_v4lsrc_try_palette):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get):
- * sys/v4l2/v4l2-overlay_calls.c: (gst_v4l2_set_display),
- (gst_v4l2_set_window), (gst_v4l2_enable_overlay):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_get_norm),
- (gst_v4l2_set_norm), (gst_v4l2_get_input), (gst_v4l2_set_input),
- (gst_v4l2_get_output), (gst_v4l2_set_output),
- (gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
- (gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
- (gst_v4l2_set_attribute):
- * sys/v4l2/v4l2_calls.h:
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
- (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
- (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
- (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
- (gst_v4l2src_capture_stop):
- * sys/vcd/vcdsrc.c: (vcdsrc_open_file):
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_buffer_new),
- (gst_xvideosink_sinkconnect), (gst_xvideosink_chain),
- (gst_xvideosink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain):
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/volume/gstvolume.c: (gst_volume_set_volume),
- (gst_volume_get_volume), (volume_class_init), (volume_init),
- (volume_chain_int16), (volume_update_volume):
- * gst/volume/gstvolume.h:
- make code more readable by removing magic numbers
- make mixer interface export 0-100 range
- make it internally map to 0.0-1.0 range so users don't distort
- output by putting the sliders at full volume
+ gst-libs/gst/audio/TODO: Updated.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/TODO:
+ Updated.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
+ On EOS, wait till the last sample is played before posting EOS.
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/play/play.c: (gst_play_tick_callback),
- (gst_play_state_change), (gst_play_seek_to_time):
- block the tick callback for 0.5 secs after doing a seek
+ * tests/check/pipelines/theoraenc.c:
+ comment on my understanding
+ Original commit message from CVS:
+ comment on my understanding
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/play/play.c: (gst_play_new):
- check for GError
+ * common:
+ * tests/check/pipelines/theoraenc.c:
+ reformat to fit 80 chars
+ Original commit message from CVS:
+ reformat to fit 80 chars
-2004-02-01 Julien MOUTTE <julien@moutte.net>
+2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
- * gst-libs/gst/play/play.c: (gst_play_seek_to_time),
- (gst_play_new): Accepting NULL GError, blocking time tick while seeking.
- * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_init): s/sinkconnect/sink_link
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_init): s/sinkconnect/sink_link
+ gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
+ Original commit message from CVS:
+ 2006-02-01 Philippe Kalaf <burger at speedy dot org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
+ setting queue_delay to zero. Also avoid thread being started if
+ queue_delay is zero.
-2004-02-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- check for a function added in vorbis 1.1
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
- (gst_alsa_drain_audio), (gst_alsa_stop_audio):
- really start/stop clock only on PLAYING <=> PAUSED
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- remove \n from debugging lines
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
- make it work when seeking does not
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- reset on DISCONT
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start):
- start clock on PAUSED=>PLAYING, not later
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- extract correct time for different discont formats
- (gst_alsa_sink_get_time):
- don't segfault when no format is negotiated yet, just return 0
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_handle_event), (gst_ogg_demux_push),
- (gst_ogg_pad_push):
- handle flush and discont events correctly
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event), (vorbis_dec_chain):
- handle discont events correctly
-
-2004-01-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.c: (gst_play_error_quark),
- (gst_play_error_create), (gst_play_error_plugin),
- (gst_play_pipeline_setup), (gst_play_init), (gst_play_new):
- * gst-libs/gst/play/play.h:
- add error handling during creation
- * examples/gstplay/player.c: (main):
- use new gst_play_new
-
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/theora/theoradec.c: (theora_dec_chain):
- make comments work
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_chain):
- add encoder tag, fix tag reading to be more error tolerant, change
- BITRATE to NOMINAL_BITRATE, add debugging, don't unref events after
- gst_pad_event_default.
- * gst/tags/gstvorbistag.c:
- (gst_tag_list_from_vorbiscomment_buffer):
- undefine function specific define at end of function
-
-2004-01-31 Jeremy Simon <jesimon@libertysurf.fr>
-
- * ext/flac/gstflac.c: (plugin_init):
- * ext/flac/gstflacdec.c: (gst_flacdec_class_init):
- * ext/flac/gstflacdec.h:
- * ext/flac/gstflacenc.h:
- Fix typos
-
-2004-01-30 David I. Lehn <dlehn@users.sourceforge.net>
-
- * examples/gstplay/player.c: s/gstplay.h/play.h/
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/gstplay.c:
- * gst-libs/gst/play/gstplay.h:
- * gst-libs/gst/play/play.c:
- more surgery, operation complete
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.old.c:
- * gst-libs/gst/play/play.old.h:
- after CVS surgery by moving, remove
- * gst-libs/gst/play/playpipelines.c:
- remove
-
- * gst/intfloat/float22int.c: (gst_float2_2_int_chain):
- add negotiation error
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_push):
- add some seeking debug info
- send a flush when seeking
-
-2004-01-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
+ Original commit message from CVS:
+ * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
+ Make test work again by connecting fakesinks to each decoded pad,
+ which makes the pipeline wait until each fakesink has a buffer
+ queued before going to PAUSED state. At that point we know the
+ decodebin pads are negotiated.
- * configure.ac:
- use AC_C_INLINE
- * configure.ac:
- * ext/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/theora/theoradec.c:
- add theora video decoder. Does just do simple decoding for now and
- has been tested against Theora cvs only. It only works when theora
- is compiled with --enable-static.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- always reset packetno on DISCONT
+2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Pass unhandled queries to the parent class's query function.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
+ (gst_cdda_base_src_handle_event):
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
+ Pass unhandled queries to the parent class's query function.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- Fix audio.
+2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
+ (gst_ogg_pad_src_query):
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
+ * ext/theora/theoradec.c: (theora_dec_src_query),
+ (theora_dec_sink_query):
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
+ (vorbis_dec_sink_query):
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
+ (gst_vorbisenc_sink_query):
+ * gst/adder/gstadder.c: (gst_adder_query):
+ Pass unhandled queries upstream instead of just
+ dropping them (#326447). Also, fix supported
+ query types list for some elements.
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegaudioparse/gstmpegaudioparse.c:
- (mp3_type_frame_length_from_header):
- Fix header parsing - stolen from ffmpeg (thank you! :) ).
+ gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (au_type_find),
+ (paris_type_find), (ilbc_type_find), (plugin_init):
+ Fix typefinding for audio/x-au, audio/x-paris and
+ audio/iLBC-sh. We cannot use the START_WITH macros
+ here, because there can only be one typefind factory
+ with the same name (caps), so the second one would
+ replace the first one and the first one would never
+ be called when doing typefinding (see #161712).
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
+ (vorbis_handle_header_packet), (vorbis_dec_push),
+ (vorbis_handle_data_packet):
+ Use scale_int when we can, add some more scaling.
+ Check packettype before parsing it.
+
+2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Call right _scale functions.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (_theora_granule_time),
+ (theora_dec_src_convert), (theora_dec_sink_convert):
+ Call right _scale functions.
+ Use parameter instead of some other random value.
+
+2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Use higher precision timestamps calculation.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (_theora_granule_frame),
+ (_theora_granule_time), (_inc_granulepos),
+ (theora_dec_src_convert), (theora_dec_sink_convert),
+ (theora_handle_type_packet), (theora_handle_data_packet),
+ (theora_dec_chain):
+ Use higher precision timestamps calculation.
+ Convert some other conversions to _scale.
+
+2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/: initialize gst_controller before using
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_create_sine_table), (plugin_init):
+ * gst/volume/gstvolume.c: (plugin_init):
+ initialize gst_controller before using
- * ext/esd/esdsink.c: (gst_esdsink_init):
- Since we have static pad template caps, we don't need to negotiate;
- either the core errors out or we know the format.
+2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c:
+ * tests/check/pipelines/vorbisenc.c:
+ Define constant using G_GINT64_CONSTANT to avoid errors when
+ passing it around - otherwise it gets truncated to 32 bits.
+ Fixes failing tests.
+
+2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
+
+ sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
+ Original commit message from CVS:
+ 2006-01-31 Andy Wingo <wingo@pobox.com>
+ * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
+ caps being set doesn't have a framerate value. Basically a stopgap
+ measure.
+ * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
+ technically correct enough to put into core though.
+ (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
+ DURATION. Fixes theoraenc ! oggmux.
+ * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
+ fraction, not double.
+
+2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * gst-plugins-base.spec.in:
+ update with latest files
+ Original commit message from CVS:
+ update with latest files
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_seek):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_seek):
- Fix event handling.
+2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-01-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ win32/vs7: add vs7 project files created by Sergey Scobich
+ Original commit message from CVS:
+ * win32/vs7:
+ add vs7 project files created by Sergey Scobich
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps):
- removee video/x-theora from vp3 decoder, it doesn't handle raw
- theora streams
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init):
- fix bug with finalizing element that never went to PAUSED
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
- length and position queries were swapped
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_from_granulepos), (vorbis_dec_src_query),
- (vorbis_dec_src_event):
- implement querying time and bytes
+2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ win32/vs8: add vs8 project files created by Sergey Scobich
+ Original commit message from CVS:
+ * win32/vs8:
+ add vs8 project files created by Sergey Scobich
- * just about every source file:
- gst_element_error -> GST_ELEMENT_ERROR
+2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
-2004-01-29 Julien MOUTTE <julien@moutte.net>
+ ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
+ timestamp + duration, not just timestamp -- ogg pages should be
+ ordered by stop time. Necessary fix given the change in vorbis
+ timestamps.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get): Fixing seeking
- emiting FLUSH and even before DISCONT.
- * gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): Fix seeking to
- get the best instant seeking as possible yay!
+2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
-2004-01-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ * tests/check/pipelines/theoraenc.c:
+ ext/theora/theoraenc.c (theora_enc_sink_setcaps)
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
+ (gst_theora_enc_init): Pull the granule shift out of the encoder.
+ (granulepos_add): New function, handles the messiness of adjusting
+ granulepos values.
+ (theora_buffer_from_packet):
+ (theora_enc_chain):
+ (theora_enc_sink_event): Use granulepos_add, not +.
+ * tests/check/pipelines/theoraenc.c
+ (check_buffer_granulepos_from_starttime): Just check the frame
+ count, not the actual granulepos -- we can't dictate to the
+ encoder when it should be placing keyframes.
+
+2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
+ SERVICE_NOT_AVAILABLE happens for example when you're trying to
+ play an http:// stream from a server that's not serving
- * gst/mpeg1videoparse/gstmp1videoparse.c:
- (gst_mp1videoparse_real_chain):
- Committed wrong version last week... Grr... Didn't notice until now.
+2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
+ * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
+ remove the UINT64_CONSTANT macro, doesn't appear to be needed or
+ available.
+
+2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
+ although theoraenc was timestamping correctly. Added handling of
+ streams that start with nonzero timestamps.
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/theoraenc.c: New file, basically does same
+ tests as vorbisenc.
+ * tests/check/pipelines/vorbisenc.c: I claim these bugs.
+
+2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
+ (gst_audioringbuffer_pause):
+ Implement pause that does not wait for completion.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Don't drop buffers when going to PAUSED but perform preroll on
+ remaining samples now that core base class supports this.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
+ (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
+ (gst_ring_buffer_commit):
+ Pause should not signal waiters.
+ Implement return value of _commit correctly.
+
+2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
+ * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
+ updated to timestamp from the first sample, not the last.
+ (gst_vorbisenc_buffer_from_header_packet): New function, takes
+ special care of granulepos and timestamp for header packets.
+ (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
+ when the first buffer has a nonzero timestamp.
+ * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
+ (GstVorbisEnc.subgranule_offset): New members. Take care of the
+ case when the first audio buffer we get has a nonzero timestamp.
+ (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
+ properly timestamp vorbis buffers with the time of the first
+ sample, not the last.
+ * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
+ vorbis_granule_time_copy -- now it takes the granule/subgranule
+ offset into account.
+ * tests/check/pipelines/vorbisenc.c: New test for correctness of
+ timestamps, durations, and granulepos on buffers produced by
+ vorbisenc.
+
+2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
+ Patch from Eric Jonas to support conversions to/from UYVY
+ (Fixes: #324626)
-2004-01-29 Julien MOUTTE <julien@moutte.net>
+2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/: Implement subtitles.
+ Original commit message from CVS:
+ 2006-01-30 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_overrun),
+ (setup_subtitle), (setup_source), (set_active_source):
+ * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
+ (gen_text_element), (gen_audio_element), (gen_vis_element),
+ (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
+
+2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
+ * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
+ use of gst_guint64_to_gdouble to be compliant with vs6
+ * gst/playback/gstdecodebin.c: (try_to_link_1)
+ * gst/videorate/videorate.c: (gst_video_rate_blank_data)
+ use of G_GINT64_CONSTANT for int64 constants
+ * win32/common/libgstinterfaces.def:
+ export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
+ * win32/vs6:
+ update and add new project files
+
+2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new): Emit the
- have_xwindow_id signal in xwindow_create.
+ add a win32-update rule like in core, and copy over enumtypes files
+ Original commit message from CVS:
+ * Makefile.am:
+ * win32/MANIFEST:
+ * win32/common/interfaces-enumtypes.c:
+ (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
+ (gst_mixer_track_flags_get_type),
+ (gst_tuner_channel_flags_get_type):
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ (gst_audio_channel_position_get_type):
+ * win32/common/multichannel-enumtypes.h:
+ add a win32-update rule like in core, and copy over enumtypes files
-2004-01-29 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c:
- lots of changes - mainly support for chained bitstreams, seeking,
- querying and bugfixes of course
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisdec.h:
- add vorbisdec raw vorbis decoder
- * ext/vorbis/vorbis.c: (plugin_init):
- register vorbisdec as PRIMARY, vorbisfile as SECONDARY
- * gst/intfloat/Makefile.am:
- * gst/intfloat/float22int.c:
- * gst/intfloat/float22int.h:
- * gst/intfloat/gstintfloatconvert.c: (plugin_init):
- add float2intnew plugin. It converts multichannel interleaved float to
- multichannel interleaved int. The name should probably be changed.
- * gst/typefind/gsttypefindfunctions.c: (theora_type_find),
- (plugin_init):
- add typefinding for raw theora video so oggdemux can detect it.
-
-2004-01-28 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): seek on video
- sink element first.
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event): Fixing src event handler.
-
-2004-01-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
- (gst_v4lsrc_open), (gst_v4lsrc_close),
- (gst_v4lsrc_palette_to_caps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_set_clock):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_start),
- (gst_v4lsrc_grab_frame), (gst_v4lsrc_capture_stop):
- Implement resizing... Hack. But that's why v4l is b0rked...
+ * configure.ac:
+ generate win32/common/config.h
+ Original commit message from CVS:
+ generate win32/common/config.h
-2004-01-28 Julien MOUTTE <julien@moutte.net>
+2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_ximage_put), (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_xwindow_destroy),
- (gst_xvimagesink_xwindow_resize), (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_xcontext_get): Removing some useless debugs messages,
- correctly cleaning the image created to check xshm calls on succes,
- added a lot of XSync calls in X11 functions, and fixed a segfault when
- no image format was defined before negotiation happened.
+ win32/: add config files just like in core
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ * win32/common/config.h.in:
+ add config files just like in core
-2004-01-28 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_query_func):
- use gst_element_get_time to get correct time
+ ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
+ (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
+ (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
+ (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
+ (gst_alsasrc_unprepare), (gst_alsasrc_read):
+ Update all error messages. All of them should either use
+ the default translated message, or actually provide a
+ translatable string.
+ Make the string for channel count problems meaningful.
-2004-01-28 Julien MOUTTE <julien@moutte.net>
+2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xerror),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xerror),
- (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_destroy), (gst_xvimagesink_xcontext_get): Our
- X plugins are now able to detect that XShm calls will fail even if the
- server claims that it has XShm support (remote displays most of the
- time). We then log the error as a GST_DEBUG and set use_shm to FALSE
- so that we use non XShm functions. This feature is almost useless for
- xvimagesink as Xv is not supported on remote displays anyway, but
- it might happen than even on the local display XShm calls fail.
+ gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
+ Make gcc-4.1 happy (part of #327357).
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_init),
- (gst_esdsink_link), (gst_esdsink_get_time), (gst_esdsink_chain),
- (gst_esdsink_change_state): Fix sync issues in esdsink. Also
- changed esdsink to only use 44100,16,2, since esd sucks at rate
- conversion and esdsink has had difficulty negotiating.
+ sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
+ Original commit message from CVS:
+ * sys/v4l/v4l_calls.c: (gst_v4l_open):
+ check for and throw RESOURCE_BUSY
-2004-01-27 Julien MOUTTE <julien@moutte.net>
+2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
- * gst-libs/gst/play/gstplay.c: (gst_play_tick_callback),
- (gst_play_seek_to_time): Fixing the way to get current position.
+ gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
+ Original commit message from CVS:
+ * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
+ checked in this change -- it requires liboil features not
+ in 0.3.6. Revert parts.
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
- * sys/oss/gstosssink.c: (gst_osssink_sink_query):
- use gst_element_get_time to get correct time
+ update liboil requirement to 0.3.6
+ Original commit message from CVS:
+ * REQUIREMENTS:
+ * configure.ac: update liboil requirement to 0.3.6
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/vs_scanline.c: liboilify
-2004-01-27 Julien MOUTTE <julien@moutte.net>
+2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/play/gstplay.c: (gst_play_set_location): The easiest
- fix ever... Inverting 2 lines of code make spider autoplug correctly
- tagged mp3 !
+ ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (get_buffer):
+ When pad_alloc returns a GstFlowReturn other
+ than GST_FLOW_OK, make sure it is passed upstream.
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Use gst_pad_try_set_caps_nonfixed().
+ ext/alsa/gstalsasink.c: Free the device name string.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
+ (gst_alsasink_class_init):
+ Free the device name string.
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
+ (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
+ (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
+ Don't remove a pad from the collectpads structure until it
+ is released - it's a request pad, and may receive data again
+ if the element gets moved back to PLAYING state.
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ Ensure we turn on double buffering on the Xv port, and
+ set the colour key to something dark and mysterious that
+ isn't black.
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/: - a library should not call setlocale. see Libraries node in gettext manual
+ Original commit message from CVS:
+ * ext/alsa/gstalsaplugin.c: (plugin_init):
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_base_init), (plugin_init):
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
+ - a library should not call setlocale. see Libraries node in
+ gettext manual
+ - make sure all plugins that use translation do bindtextdomain
+ to point to the localedir
+ * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
+ (setup_sinks), (plugin_init):
+ all this, and check for NULL when creating sinks
- * gst/ac3parse/gstac3parse.c: update to checklist 5
- * gst/adder/gstadder.c: rewrite negotiation. update to checklist 5
- * gst/audioconvert/gstaudioconvert.c: update to checklist 5
- * gst/audioscale/gstaudioscale.c: same
- * gst/auparse/gstauparse.c: same
- * gst/avi/gstavidemux.c: same
+2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
+ Original commit message from CVS:
+ 2006-01-27 Julien MOUTTE <julien@moutte.net>
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
+ (plugin_init): Make typefinding of subtitles work again.
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
- stop processing after EOS
+2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
+ (mp3_type_frame_length_from_header), (mp3_type_find),
+ (wavpack_type_find), (m4a_type_find), (ircam_type_find),
+ (plugin_init):
+ Backport a bunch of typefinding fixes from the 0.8 branch.
+ Also, improve wavpack typefinding: if we can't peek the
+ entire wavpack block, try to parse the bits we can get and
+ see if we find what we're looking for in those.
- * gst/asfdemux/asfheaders.h:
- * gst/asfdemux/gstasfdemux.c:
- * gst/asfdemux/gstasfmux.c: (gst_asfmux_put_guid),
- (gst_asfmux_put_string), (gst_asfmux_put_wav_header),
- (gst_asfmux_put_vid_header), (gst_asfmux_put_bmp_header):
- lot's of fixes to make data extraction simpler and get the code
- architecture and compiler independant. Add debugging category
- * gst/goom/gstgoom.c: (gst_goom_change_state):
- reset channel count on PAUSED=>READY, not READY=>PAUSED
+2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Handle some more cases of pixel aspect ratio.
+ Original commit message from CVS:
+ 2006-01-26 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c:
+ (gst_ximagesink_calculate_pixel_aspect_ratio):
+ * sys/xvimage/xvimagesink.c:
+ (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
+ more cases of pixel aspect ratio.
+
+2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (pad_probe):
+ Also consider the flush-start and tag events as unblockers
+ for the pad probes.
+
+2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
+ Original commit message from CVS:
+ 2006-01-26 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybin.c: (gst_play_bin_init),
+ (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
+ (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
+ On the fly visualisation switch, works disabling, enabling as
+ well but it won't be able to enable vis in a playbin that was
+ created with no visualisation.
+
+2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Undo previous commit, it breaks resume after pause.
+
+2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
+ Improve debugging.
+ Post error when caps cannot be parsed.
+ Resync on discontinuity in the stream.
+ Clip samples to segment boundaries.
+ return WRONG_STATE sooner when we are flushing.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create):
+ Make audiosrc operate in TIME.
+ Set TIMESTAMP and DURATION on buffers.
+
+2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/examples/seek/seek.c: Output tag messages as well.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (main):
+ Output tag messages as well.
-2004-01-26 Colin Walters <walters@verbum.org>
+2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
+ (free_pad_probes), (remove_fakesink), (pad_probe),
+ (close_pad_link), (gst_decode_bin_change_state):
+ Replace GstPadBlockCallback with pad probes that detect
+ first buffer AND eos before removing fakesink.
+ Fixes hang with demuxers doing EOS while pre-rolling.
+ Solves #328279
+
+2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
+
+ GCC 2.95 fixes (#328263).
+ Original commit message from CVS:
+ 2006-01-23 Andy Wingo <wingo@pobox.com>
+ * ext/alsa/gstalsasink.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_add_to_queue),
+ (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
+ Patch by: Jens Granseuer <jensgr at gmx dot net>
+
+2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
+ Original commit message from CVS:
+ 2006-01-22 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
+ frames. We might get a frame destroyed after changing state to
+ NULL, adding a safety check on xcontext.
+
+2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ Fix prepare-xwindow-id code example in the docs - we need to
+ ignore all messages that aren't element messages as well.
- * ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_get): Remove ugly
- code to pull a bigger buffer in iradio mode. This as a side effect
- makes typefinding work.
+2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
+ sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
+ Original commit message from CVS:
+ 2006-01-21 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
+ I think one day i'll completely undestand how caps negotiation
+ is supposed to work. This refactoring handles buffer_alloc
+ called with caps we can't handle. We definitely don't want a
+ set_caps with those caps, so we define and allocate a buffer
+ we would like to receive.
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_extradata):
- Fix SVQ3 decoding on PPC
+2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-26 Julien MOUTTE <julien@moutte.net>
+ * autogen.sh:
+ * common:
+ up automake requirement to 1.7
+ Original commit message from CVS:
+ up automake requirement to 1.7
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Dunno how
- that one managed to stay there... Fixed.
+2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
+ gst/playback/gstplaybasebin.c: Free iterator when done.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Free iterator when done.
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
- (gst_ffmpeg_caps_to_extradata), (gst_ffmpeg_caps_to_pixfmt):
- * gst/qtdemux/qtdemux.c: (plugin_init), (qtdemux_parse_trak),
- (qtdemux_video_caps):
- * gst/qtdemux/qtdemux.h:
- Add SVQ3 specific flags to qtdemux and ffmpeg
+2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Fix playback of non-synchronised streams by assuming a rate
+ of 1.0 instead of a random one.
+ Makes this work again:
+ gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
+ endianness=(int)4321, signed=(boolean)true, width=(int)16,
+ depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
+ audioresample ! alsasink
- * gst-libs/gst/audio/audio.h:
- remove buffer-frames from audio caps
- * gst/audioconvert/gstaudioconvert.c:
- fix plugin to really work.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/mixer/mixer.c:
- * gst-libs/gst/propertyprobe/propertyprobe.c:
- * gst-libs/gst/tuner/tuner.c: (gst_tuner_find_norm_by_name),
- (gst_tuner_find_channel_by_name):
- * gst-libs/gst/tuner/tuner.h:
- Add gtk-doc style comments. Also fix a function name.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c: (gst_divxdec_init),
- (gst_divxdec_negotiate):
- Fix for new capsnego - also fixes gst-player with divxdec.
-
-2004-01-25 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_identity_handoff), (gst_play_set_location),
- (gst_play_set_visualization), (gst_play_connect_visualization): Another
- try in visualization implementation. Still have an issue with switch
- blocking when pulling from video_queue and only audio comes out of
- spider.
- * gst/switch/gstswitch.c: (gst_switch_release_pad),
- (gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
- release method. And check if the pad is usable before pulling.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose),
- (gst_videobalance_init),
- (gst_videobalance_colorbalance_list_channels),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value),
- (gst_videobalance_update_properties),
- (gst_videobalance_update_tables_planar411),
- (gst_videobalance_planar411):
- * gst/videofilter/gstvideobalance.h:
- Implement lookup-tables. +/- 10x faster.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
- The index reading was broken. The rest worked fine, but the whole
- goal of my rewrite was to make avidemux readable, and this was
- not at all readable. Please use typed variables.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- Additional pad usability check.
- * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
- (mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
- (gst_mp1videoparse_real_chain):
- Fix MPEG video stream parsing. The original plugin had several
- issues, including not timestamping streams where the source was
- not timestamped (this happens with PTS values in mpeg system
- streams, but MPEG video is also a valid stream on its own so
- that needs timestamps too). We use the display time code for that
- for now. Also, if one incoming buffer contains multiple valid
- frames, we push them all on correctly now, including proper EOS
- handling. Lastly, several potential segfaults were fixed, and we
- properly sync on new sequence/gop headers to include them in next,
- not previous frames (since they're header for the next frame, not
- the previous). Also see #119206.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
- (bpf_from_header):
- Move caps setting so we only do it after finding several valid
- MPEG-1 fraes sequentially, not right after the first one (which
- might be coincidental).
- * gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (plugin_init):
- Add unsynced MPEG video stream typefinding, and change some
- probability values so we detect streams rightly. The idea is as
- follows: I can have an unsynced system stream which contains
- video. In the current code, I would randomly get a type for either
- system or video stream type found, because the probabilities are
- being calculated rather randomly. I now use fixed values, so we
- always prefer system stream if that was found (and that is how it
- should be). If no system stream was found, we can still identity the stream as video-only.
-
-2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
- don't write to buffer. Extract data without the need of
- __attribute__ ((packed))
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (mpeg1_parse_header),
- (mpeg1_sys_type_find):
- Fix MPEG-1 stream typefinding.
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Fix typefinding for MPEG-1 system streams, similar to MPEG-2.
-
-2004-01-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/esd/esdsink.c: (gst_esdsink_open_audio):
- * ext/esd/gstesd.c: (plugin_init):
- private debugging, better error reporting
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
- (gst_riff_read_init), (gst_riff_read_change_state):
- * gst-libs/gst/riff/riff-read.h:
- Remove stuff fromold metadata system.
+2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * ext/ogg/gstoggdemux.c:
- Fix wrong file comment.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
- Add metadata reading properly.
+=== release 0.10.2 ===
-2004-01-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/Makefile.am:
- Fix nas DIST_SUBDIRS
- Uraeus:
- Fix bug where make distcheck doesn't get run on adding stuff to
- the build.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.2
+ Original commit message from CVS:
+ releasing 0.10.2
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/divx/gstdivxdec.c: (gst_divxdec_init), (gst_divxdec_setup):
- * ext/divx/gstdivxdec.h:
- Fix divx3 ("msmpeg4") playback using divxdec.
+ gst/playback/: Comment out broken code that connects to the state-changed signal.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property):
+ Comment out broken code that connects to the state-changed signal.
+ At this point, changing current stream selection is broken, but
+ stuff like gst-launch playbin current-audio=1 works and filters
+ to the chosen stream.
-2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/typefind/gsttypefindfunctions.c:
- (mp3_type_frame_length_from_header): fix bug in length computation
- (mp3_type_find): improve debugging output
-
-2004-01-23 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_set_location), (gst_play_seek_to_time),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_sink_element): Reworked
- the pipeline from scratch. Visualization is back and switch went out as
- i realized it was not possible to use the way i wanted.
- * sys/ximage/ximagesink.c: (gst_ximagesink_imagepool_clear),
- (gst_ximagesink_change_state), (gst_ximagesink_dispose): Move xcontext
- clearing in state change from READY to NULL. So that one can clean the
- X ressources keeping the element.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_imagepool_clear), (gst_xvimagesink_change_state),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_set_property), (gst_xvimagesink_dispose),
- (gst_xvimagesink_init): Same xcontext cleaning than ximagesink in state
- change from READY to NULL and fixed some stupid bugs in colorbalance
- get/set values. Also added the following feature : when nobody tries to
- set some values to the colorbalance levels before the xcontext is
- grabbed, then when creating channels list from Xv attributes we set the
- internal values to the Xv defaults. This way we handle buggy Xv drivers
- that set default hue values far from the middle of the range (Thanks
- to Jon Trowbridge for pointing that issue).
- * sys/xvimage/xvimagesink.h: Adding a cb_changed boolean to know if
- colorbalance levels have been set before xcontext is grabbed.
-
-2004-01-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Fix the ossmixer case where we shouldn't open /dev/dsp* because
- it might block operations (which is bad for a mixer).
-
-2004-01-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/media-info/media-info-priv.c: (have_type_callback),
- (deep_notify_callback), (gmi_set_decoder), (gmi_clear_decoder),
- (gmip_find_type_pre):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info.c:
- (gst_media_info_instance_init), (gst_media_info_read_idler):
- add fakesink to get caps on decoder src pad again
- fix callback prototype to match new have_type signal signature
-
-2004-01-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/adder/gstadder.c: (gst_adder_link):
- fix non-compile and cut-n-paste code
-
-2004-01-21 David Schleef <ds@schleef.org>
-
- * ext/swfdec/gstswfdec.c: (gst_swfdec_video_getcaps),
- (gst_swfdec_video_link), (copy_image), (gst_swfdec_loop),
- (gst_swfdec_init), (gst_swfdec_change_state):
- * ext/swfdec/gstswfdec.h:
- Fix negotiation.
- * gst/adder/gstadder.c: (gst_adder_link), (gst_adder_init),
- (gst_adder_request_new_pad): Fix negotiation.
- * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_fixate):
- Add a fixate function.
- * gst/intfloat/gstfloat2int.c:
- * gst/intfloat/gstfloat2int.h:
- * gst/intfloat/gstint2float.c:
- * gst/intfloat/gstint2float.h:
- Completely rewrite the negotiation. Doesn't quite work yet,
- due to some buffer-frames problem.
-
-2004-01-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
+ Fix #327216 (null dereference in vorbisdec)
- * ext/gnomevfs/gstgnomevfssrc.c:
- * sys/v4l2/v4l2_calls.h:
- fix includes for distcheck
+2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-21 Christian Schaller <uraeus@gnome.org>
+ ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_comment_packet):
+ Post taglist actually on bus instead of just freeing it
+ (fixes #327114 and totem bug #327080).
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
+ Use gst_element_found_tags_for_pad(), so that the tags
+ are sent downstream as an event as well.
- * ext/nas/
- Add libnas (network audio system) plugin, patch from Arwed von Merkatz
- based on earlier patch from Laurent Vivier
+2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Jeremy Simon <jesimon@libertysurf.fr>
+ sys/: move all regularly occurring messages to GST_LOG level add some more object logs
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
+ (gst_ximagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
+ (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
+ (gst_xvimagesink_buffer_alloc):
+ move all regularly occurring messages to GST_LOG level
+ add some more object logs
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_extradata):
- Fix wma caps property
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Fix typo (flags1 and flags2)
+2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ prerelease
+ Original commit message from CVS:
+ prerelease
- * gst-libs/gst/media-info/media-info-priv.c:
- (deep_notify_callback), (gmi_seek_to_track), (gmi_get_decoder),
- (gmip_find_type_pre), (gmip_find_type), (gmip_find_stream_post),
- (gmip_find_stream), (gmip_find_track_metadata),
- (gmip_find_track_streaminfo_post), (gmip_find_track_streaminfo),
- (gmip_find_track_format):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c: (main):
- * gst-libs/gst/media-info/media-info.c: (gst_media_info_init),
- (gst_media_info_read_idler), (gst_media_info_read):
- * gst-libs/gst/media-info/media-info.h:
- register debugging category and use it for debugging
+2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/ogg/gstoggmux.c: fix a silly segfault
+ Original commit message from CVS:
+ 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
+ fix a silly segfault
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_update_streaminfo),
- (gst_vorbisfile_new_link):
- signal streaminfo through tags
+2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ Add docs for mixerutils stuff.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/mixerutils.c:
+ * gst-libs/gst/audio/mixerutils.h:
+ Add docs for mixerutils stuff.
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- g++ doesn't like NULL in our i18n/error macros, should be
- either (NULL) or ("").
+2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Fix playback for sources that emit raw audio or
+ raw video streams (e.g.: cd audio sources) (#325984).
- * sys/dxr3/dxr3audiosink.c:
- * sys/dxr3/dxr3init.c:
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_close):
- Fix more error error error errors (missing includes here).
+2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/mixerutils.c: actually save the element we create
+ Original commit message from CVS:
+ * gst-libs/gst/audio/mixerutils.c:
+ (gst_audio_mixer_filter_do_filter):
+ actually save the element we create
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- fix thomas' error errors.
+2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-plugins-base.spec.in:
+ remove version suffix
+ Original commit message from CVS:
+ remove version suffix
- * ext/mpeg2enc/gstmpeg2enc.cc:
- fix error errors.
+2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_handle_track_seek):
+ No need to post a tag message on the bus when seeking
+ within the same track, only post it when the current
+ track changes.
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- Fix for new error system.
+2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_destroy),
+ (probe_triggered), (new_decoded_pad), (mute_group_type),
+ (set_active_source):
+ * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_base_init),
+ (gst_stream_selector_set_property),
+ (gst_stream_selector_request_new_pad):
+ Reenable stream selection. These mechanisms need a complete overhaul
+ in the face of 0.8->0.10 changes though.
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- fix for new error reporting
+2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-20 David Schleef <ds@schleef.org>
+ ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c:
+ Change the pad template to src_%d to match the pads that
+ are created from it. decodebin needs this information in order
+ to decide that oggdemux is capable of producing multiple pads
+ (and hence needs queues inserted).
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
+ (gst_ogg_mux_collected):
+ Make debug output more useful by using GST_PTR_FORMAT.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_getcaps),
- (gst_ximagesink_set_xwindow_id): Change to using a framerate
- of [1,100] instead of [0,MAX], since 0 isn't handled correctly,
- and neither is 100+, most likely.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_getcaps): same
+2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * gst-plugins-base.spec.in:
+ update spec.in file
+ Original commit message from CVS:
+ update spec.in file
- * configure.ac:
- Up version requirement to 2.0.3 (not yet released) to avoid symbol
- clashes with ffmpeg.
+2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Julien MOUTTE <julien@moutte.net>
+ gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
+ Original commit message from CVS:
+ Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
+ Set depth and width for alaw/mulaw (fixes #326601).
- * gst/switch/gstswitch.c: (gst_switch_request_new_pad),
- (gst_switch_init): Fixed switch element : proxying link and setting
- caps from src to sink on request.
+2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ tests/icles/Makefile.am: don't build the tests if we don't have the libs
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ don't build the tests if we don't have the libs
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- * sys/v4l2/v4l2-overlay_calls.c: (gst_v4l2_set_display),
- (gst_v4l2_set_window), (gst_v4l2_enable_overlay):
- fix element_error
+2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
+ (gst_cd_paranoia_paranoia_callback):
+ Don't try to free NULL pointers.
- * sys/v4l/v4l_calls.h:
- * sys/v4l2/v4l2_calls.h:
- element_error fixes
+2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/audiorate/gstaudiorate.c: Add debugging category.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
+ (gst_audio_rate_change_state), (plugin_init):
+ Add debugging category.
+ Fix type issues.
+ Add case for incoming buffers without valid offset/offset_end.
- * gst-libs/gst/gst-i18n-plugin.h:
- add locale.h
- remove config.h inclusion
+2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
+ Don't leak GCond in audio sources.
- * autogen.sh:
- adding autopoint invocation
- * Makefile.am:
- * configure.ac:
- * gst-libs/gst/gettext.h:
- adding gettext bits
- * ext/audiofile/gstafsink.c: (gst_afsink_plugin_init):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_open_file):
- * ext/sndfile/gstsf.c: (gst_sf_loop), (plugin_init):
- * gst-libs/gst/gst-i18n-plugin.h:
- * gst/avi/gstavi.c: (plugin_init):
- * sys/dxr3/dxr3init.c: (plugin_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_write_data):
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- * sys/v4l2/gstv4l2.c: (plugin_init):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_get_norm), (gst_v4l2_set_norm),
- (gst_v4l2_get_input), (gst_v4l2_set_input), (gst_v4l2_get_output),
- (gst_v4l2_set_output), (gst_v4l2_get_frequency),
- (gst_v4l2_set_frequency), (gst_v4l2_signal_strength),
- (gst_v4l2_get_attribute), (gst_v4l2_set_attribute):
- make sure locale and translation domain are set
- fix translated strings
- * po/.cvsignore:
- * po/LINGUAS:
- * po/Makevars:
- * po/POTFILES.in:
- * po/nl.po:
- put translation files into place
- * sys/xvideo/imagetest.c: (main):
- * ext/dv/demo-play.c: (main):
- fix unnecessary translations
+2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-19 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_audio_element):
+ Don't leak an autoaudiosink/alsasink when we generate
+ a new audio element. (old code, I guess)
- * ext/sndfile/gstsf.c:
- * gst/avi/gstavimux.c:
- * ext/audiofile/gstafsink.c:
- * ext/audiofile/gstafsrc.c:
- * ext/gnomevfs/gstgnomevfssink.c:
- * ext/gnomevfs/gstgnomevfssrc.c:
- * sys/oss/gstosselement.c:
- * sys/v4l/v4l_calls.h:
- fix i18n include
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_get_norm),
- (gst_v4l2_set_norm), (gst_v4l2_get_input), (gst_v4l2_set_input),
- (gst_v4l2_get_output), (gst_v4l2_set_output),
- (gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
- (gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
- (gst_v4l2_set_attribute):
- update to new error handling
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/sidplay/gstsiddec.cc:
- * gst/modplug/gstmodplug.cc:
- parenthese NULL because C++ seems angry about it
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/gst-i18n-plugin.h:
- add skeleton i18n stuff, but needs to be further implemented
-
-2004-01-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * examples/gstplay/player.c: (main):
- * ext/aalib/gstaasink.c: (gst_aasink_open):
- * ext/alsa/gstalsa.c: (gst_alsa_link), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
- (gst_afsink_close_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
- (gst_afsrc_close_file):
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_user_op), (dvdnavsrc_get):
- * ext/esd/esdmon.c: (gst_esdmon_get):
- * ext/esd/esdsink.c: (gst_esdsink_chain):
- * ext/faac/gstfaac.c: (gst_faac_chain):
- * ext/faad/gstfaad.c: (gst_faad_chain):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_error_callback),
- (gst_flacdec_loop):
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (gst_gnomevfssrc_open_file):
- * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
- * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_chain):
- * ext/lcs/gstcolorspace.c: (gst_colorspace_srcconnect_func):
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_do_typefind), (gst_id3_tag_chain):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain),
- (gst_ogg_demux_push), (gst_ogg_pad_push):
- * ext/raw1394/gstdv1394src.c:
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_lock),
- (gst_sdlvideosink_initsdl), (gst_sdlvideosink_create):
- * ext/sidplay/gstsiddec.cc:
- * ext/sndfile/gstsf.c: (gst_sf_open_file), (gst_sf_close_file),
- (gst_sf_loop):
- * ext/speex/gstspeexenc.c: (gst_speexenc_chain):
- * ext/tarkin/gsttarkindec.c: (gst_tarkindec_chain):
- * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_loop):
- * ext/xvid/gstxviddec.c: (gst_xviddec_setup), (gst_xviddec_chain):
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_setup), (gst_xvidenc_chain):
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_element_data), (gst_riff_read_seek),
- (gst_riff_peek_list), (gst_riff_read_list), (gst_riff_read_header):
- * gst/adder/gstadder.c: (gst_adder_parse_caps), (gst_adder_loop):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_segment),
- (gst_asf_demux_process_stream), (gst_asf_demux_get_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_init),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- * gst/avi/gstavimux.c: (gst_avimux_stop_file):
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c: (gst_goom_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/intfloat/gstfloat2int.c: (gst_float2int_loop):
- * gst/intfloat/gstint2float.c: (gst_int2float_chain_gint16):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_uint), (gst_ebml_read_sint),
- (gst_ebml_read_float), (gst_ebml_read_header):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
- (gst_matroska_demux_parse_blockgroup):
- * gst/modplug/gstmodplug.cc:
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- * gst/oneton/gstoneton.c: (gst_oneton_chain):
- * gst/silence/gstsilence.c: (gst_silence_get):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/smpte/gstsmpte.c: (gst_smpte_loop):
- * gst/speed/gstspeed.c: (speed_loop):
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * gst/volenv/gstvolenv.c: (gst_volenv_chain):
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_loop):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_open),
- (dxr3audiosink_set_mode_pcm), (dxr3audiosink_set_mode_ac3),
- (dxr3audiosink_close):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_open), (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_open),
- (dxr3videosink_close), (dxr3videosink_write_data):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/oss/gstosselement.h:
- * sys/oss/gstosssink.c: (gst_osssink_get_type), (gst_osssink_init),
- (gst_osssink_chain):
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_buffer_free):
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_window),
- (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_get_capabilities), (gst_v4l_open),
- (gst_v4l_set_chan_norm), (gst_v4l_get_signal),
- (gst_v4l_get_frequency), (gst_v4l_set_frequency),
- (gst_v4l_get_picture), (gst_v4l_set_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame),
- (gst_v4lmjpegsrc_sync_next_frame), (gst_v4lmjpegsrc_set_capture),
- (gst_v4lmjpegsrc_set_capture_m), (gst_v4lmjpegsrc_capture_init),
- (gst_v4lmjpegsrc_requeue_frame):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_capture_init),
- (gst_v4lsrc_requeue_frame), (gst_v4lsrc_try_palette):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get):
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
- (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
- (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
- (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
- (gst_v4l2src_capture_stop):
- * sys/vcd/vcdsrc.c: (vcdsrc_open_file):
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_buffer_new),
- (gst_xvideosink_sinkconnect), (gst_xvideosink_chain),
- (gst_xvideosink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain):
- use new error signal, function and categories
-
-2004-01-18 Jeremy Simon <jesimon@libertysurf.fr>
+2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- * ext/Makefile.am:
- * ext/musicbrainz/gsttrm.c:
- * ext/musicbrainz/gsttrm.h:
- * ext/musicbrainz/Makefile.am:
- Add a trm plugin
+ gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
+ Support float audio in audiorate.
+ Use width rather than depth for selecting sample width.
-2004-01-18 Julien MOUTTE <julien@moutte.net>
+2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_set_property),
- (gst_ximagesink_get_property), (gst_ximagesink_class_init): Adding
- synchronous property for debugging.
- * sys/ximage/ximagesink.h: Adding the synchronous boolean flag.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_set_property): Moving a pointer declaration to a
- smaller block, fixing indent.
+ gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.h:
+ Use GLib types here (that way we don't have to include the
+ generated _stdint.h header, which makes life easier for win32
+ folks that don't use autotools for the build) (#325990, patch
+ by: Sergey Scobich).
-2004-01-16 David Schleef <ds@schleef.org>
+2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videofilter/gstvideobalance.c: Fix regression; changing a
- property affects the video stream.
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add synchronous property for debugging. Should probably be
- disabled in non-CVS builds. Make sure that the Xv attribute
- exists before we set it (crash!). Fix a silly float bug that
- caused colorbalance to just not work.
+ gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
+ (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (wait_segment):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Name (private) union, makes Forte compiler happy (this time
+ for real) (#324900).
-2004-01-17 Christian Schaller <Uraeus@gnome.org>
+2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * tools/gst-launch-ext.in - update for new plugins
+ gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ Link against libgstinterfaces, needed for mixer
+ and property probe stuff.
-2004-01-16 David Schleef <ds@schleef.org>
+2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect): Fix use of
- already-freed caps.
+ gst-libs/gst/Makefile.am:
+ Original commit message from CVS:
+ * gst-libs/gst/Makefile.am:
-2994-01-16 Christian Schaller <Uraeus@gnome.org>
+2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Update spec for new colorspace plugin and libcaca plugin
- * Fix compilation of libcaca plugin (clock -> id)
+ gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/mixerutils.c:
+ (gst_audio_mixer_filter_do_filter),
+ (gst_audio_mixer_filter_check_element),
+ (gst_audio_mixer_filter_probe_feature),
+ (element_factory_rank_compare_func),
+ (gst_audio_default_registry_mixer_filter):
+ * gst-libs/gst/audio/mixerutils.h:
+ Add gst_audio_default_registry_mixer_filter() utility
+ function.
-2004-01-16 Julien MOUTTE <julien@moutte.net>
+2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init): Implementing
- correct colorbalance properties. They can now be set when the element
- is still in NULL state. The values will be committed to the Xv Port
- when xcontext is initialized.
- * sys/xvimage/xvimagesink.h: Added hue, saturation, contrast,
- brightness int values in the GstXvImagesink structure.
+ gst/audioresample/resample.h: As before, but for o_buf
+ Original commit message from CVS:
+ * gst/audioresample/resample.h:
+ As before, but for o_buf
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/Makefile.am:
- restructure so having local patches works easier.
+ gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
+ Original commit message from CVS:
+ * gst/audioresample/resample.h:
+ Declare struct _ResampleState.buffer as unsigned char *, not void *,
+ since we do arithmetic on it.
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mpeg2enc/Makefile.am:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- Bugfix with respect to EOS handling.
+ gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
+ (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (wait_segment):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Sun's Forte compiler doesn't seem to like anonymous structs,
+ so use same setup as in GstBaseSrc (fixes #324900).
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect):
- Link with right caps (else, it segfaults).
- * ext/mplex/gstmplexjob.cc:
- Fix for slight API change in 1.6.1.93 release of mjpegtools.
+ move old example to tests/examples/volume/volune.c
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/volume/Makefile.am:
+ * gst/volume/demo.c:
+ move old example to tests/examples/volume/volune.c
+ * tests/examples/Makefile.am:
+ * tests/examples/seek/seek.c: (main):
+ change window-close event from "delete-event" to "destroy"
+ * tests/examples/volume/Makefile.am:
+ * tests/examples/volume/volume.c: (value_changed_callback),
+ (setup_gui), (message_received), (eos_message_received), (main):
+ fix event handling and bus usage
-2004-01-15 David Schleef <ds@schleef.org>
+2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
+ (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
+ (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
+ (gst_audio_test_src_create_square),
+ (gst_audio_test_src_create_saw),
+ (gst_audio_test_src_create_triangle),
+ (gst_audio_test_src_create_silence),
+ (gst_audio_test_src_create_white_noise),
+ (gst_audio_test_src_create_pink_noise),
+ (gst_audio_test_src_init_sine_table),
+ (gst_audio_test_src_create_sine_table),
+ (gst_audio_test_src_change_wave),
+ (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
+ (gst_audio_test_src_create), (gst_audio_test_src_set_property):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ update to basesrc changes, implement segmented seeking and eos handling,
+ add a 'sine-tab' waveform for performance critical playback
+
+2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ po/POTFILES.in: ... and this time the other modified file that I missed last time.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ ... and this time the other modified file that I missed last time.
- * gst-libs/gst/audio/Makefile.am:
- Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
- from the template.
- * gst-libs/gst/audio/gstaudiofilter.c:
- * gst-libs/gst/audio/gstaudiofilter.h:
- Add bytes_per_sample and size and n_samples calculation.
- * gst-libs/gst/audio/gstaudiofilterexample.c:
- Remove, now autogenerated.
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- Moved from gstaudiofilterexample, object name changed, code added
- so that it actually works.
- * gst-libs/gst/audio/make_filter:
- Script to build an audiofilter subclass from the template.
- * gst/colorspace/Makefile.am:
- * gst/colorspace/yuv2yuv.c:
- Remove file, since it's GPL, and we don't use it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): Making both of
- them use the buffer free function to test how the buffer was allocated.
-
-2004-01-15 David Schleef <ds@schleef.org>
-
- * ext/esd/esdsink.c: (gst_esdsink_class_init): Remove property
- that handles osssink fallback.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_getcaps):
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Add audio/x-qdm2 for QDM2 audio.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/sine/gstsinesrc.h: Add example of how to implement tags.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- Decrease minimum size to 16x16.
- * gst/wavparse/gstwavparse.c:
- Convert disabled pad template caps to new caps.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain): Throw element error when display cannot
- be opened. Increase minimum framerate to 1.0. Check the data
- free function on a buffer to make sure it is the type we expect
- before manipulating it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_init),
- (gst_videobalance_colorbalance_set_value): Implement passthru if
- settings are in the middle.
- * tools/gst-launch-ext.in: Stop using xvideosink, use ximagesink.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/videofilter/Makefile.am:
- * gst/volume/Makefile.am:
- Since we use videofilter symbols, link to it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init): Setting
- mixer interface type to HARDWARE.
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init): Adding a default
- type to SOFTWARE.
- * gst-libs/gst/mixer/mixer.h: Adding mixer interface type and macro.
- * gst-libs/gst/mixer/mixertrack.h: Adding mixertrack flag SOFTWARE.
- * gst/volume/gstvolume.c: (gst_volume_interface_supported),
- (gst_volume_interface_init), (gst_volume_list_tracks),
- (gst_volume_set_volume), (gst_volume_get_volume),
- (gst_volume_set_mute), (gst_volume_mixer_init),
- (gst_volume_dispose), (gst_volume_get_type), (volume_class_init),
- (volume_init): Implementing mixer interface.
- * gst/volume/gstvolume.h: Adding tracklist for mixer interface.
- * sys/oss/gstosselement.c: (gst_osselement_get_type),
- (gst_osselement_change_state): Removing some trailing commas in
- structures.
- * sys/oss/gstossmixer.c: (gst_ossmixer_interface_init): Setting mixer
- interface type to HARDWARE.
- * sys/v4l/gstv4lcolorbalance.c:
- (gst_v4l_color_balance_interface_init): Setting colorbalance interface
- type to HARDWARE.
- * sys/v4l2/gstv4l2colorbalance.c:
- (gst_v4l2_color_balance_interface_init): Setting colorbalance
- interface type to HARDWARE.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use exactly the
- same code than ximagesink for event handling.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/snapshot/Makefile.am:
- * ext/snapshot/gstsnapshot.c: (gst_snapshot_sinkconnect),
- (gst_snapshot_chain):
- * ext/snapshot/gstsnapshot.h:
- This has to be a joke... Snapshot should be connected to a tee,
- colorspace element before it and EOS after that, where the other
- src of the tee receives normal data.
- The current way is *wrong*.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/hermes/gsthermescolorspace.c:
- Fix another compile error. Same as below.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/colorspace/gstcolorspace.c:
- * gst/colorspace/yuv2yuv.c: (gst_colorspace_yuy2_to_i420),
- (gst_colorspace_i420_to_yv12):
- Fix compiling... Didn't test if it actually works.
-
-2004-01-15 David Schleef <ds@schleef.org>
+2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- * gst/colorspace/Makefile.am:
- * gst/colorspace/gstcolorspace.c:
- * gst/colorspace/gstcolorspace.h:
- * gst/colorspace/yuv2rgb.c:
- * gst/colorspace/yuv2rgb.h:
- Duplicate the ext/hermes colorspace plugin, and remove Hermes
- code and GPL code. Fix for new caps negotiation. Rewrite
- much of the format handling code, and some of the conversion
- code. Basically, rewrote almost everything. This element
- handles I420, YV12 to RGB conversions.
- * ext/hermes/Makefile.am:
- * ext/hermes/gsthermescolorspace.c:
- Rename colorspace to hermescolorspace. Fix negotiation issues.
- Remove non-Hermes related code. This element handles lots of
- RGB to RGB conversions, but no YUV.
- * ext/hermes/gstcolorspace.c:
- * ext/hermes/gstcolorspace.h:
- * ext/hermes/rgb2yuv.c:
- * ext/hermes/yuv2rgb.c:
- * ext/hermes/yuv2rgb.h:
- * ext/hermes/yuv2rgb_mmx16.s:
- * ext/hermes/yuv2yuv.c:
- * ext/hermes/yuv2yuv.h:
- Remove old code.
-
-2004-01-14 Colin Walters <walters@verbum.org>
-
- * ext/mad/gstid3tag.c (gst_id3_tag_chain): Don't nego caps if
- they've already been.
-
-2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_caps_nego):
- assume tag mode when pad is not connected
-
-2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Don't update the time of the clock
- (gst_alsa_sink_loop):
- sync to the clock given to alsasink, not the own clock
- * sys/oss/gstosssink.c: (gst_osssink_chain):
- sync to the clock
- (gst_osssink_change_state):
- activate the clock
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
- remove bogus code that made DISCONT events unhandled
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
- explicitly case to double in _set_simple. (fixes 2nd warning in bug
- #131502)
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_read_object_header),
- (gst_asf_demux_handle_sink_event), (gst_asf_demux_audio_caps),
- (gst_asf_demux_add_audio_stream), (gst_asf_demux_video_caps):
- convert g_warning because of wrong asf data to GST_WARNINGs (fixes
- 2nd warning in bug #131502)
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_init),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value): Fixing videobalance ranges
- for colorbalance interface implementation.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_set_property), (gst_ximagesink_get_property),
- (gst_ximagesink_dispose), (gst_ximagesink_init),
- (gst_ximagesink_class_init): Adding DISPLAY property.
- * sys/ximage/ximagesink.h: Adding display_name to store display.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_dispose), (gst_xvimagesink_init),
- (gst_xvimagesink_class_init): Adding DISPLAY property and colorbalance
- properties (they still need polishing though for gst-launch use : no
- xcontext yet, i ll do that tomorrow).
- * sys/xvimage/xvimagesink.h: Adding display_name to store display.
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_set_location), (gst_play_set_visualization): Preparing
- switch integration, adding videobalance in the pipeline.
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalance.c:
- (gst_color_balance_class_init): Adding a default type.
- * gst-libs/gst/colorbalance/colorbalance.h: Adding a macro to access
- the type.
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_get_type),
- (gst_videobalance_dispose), (gst_videobalance_class_init),
- (gst_videobalance_init), (gst_videobalance_interface_supported),
- (gst_videobalance_interface_init),
- (gst_videobalance_colorbalance_list_channels),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value),
- (gst_videobalance_colorbalance_init): Implementing colorbalance
- interface.
- * gst/videofilter/gstvideobalance.h: Adding colorbalance channels
- list.
- * sys/ximage/ximagesink.c: (gst_ximagesink_set_xwindow_id): Fixing a
- bug which was triggering a BadAccess X error when setting an overlay
- before pad was really negotiated.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_colorbalance_init):
- Using the colorbalance type macro.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/flac/gstflacenc.c: (gst_flacenc_set_metadata),
- (gst_flacenc_chain):
- handle tags correctly
- * gst/tags/gstid3tag.c: (gst_tag_list_new_from_id3v1):
- extract ID3v1 tags correctly
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (matroska_type_find),
- (plugin_init):
- Improve matroska typefinding for odd-typed headers...
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
- Fix for using incremental number on padnames.
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- Set category to divx4linux instead of divx (too generic).
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt), (gst_wavparse_handle_sink_event),
- (gst_wavparse_loop), (gst_wavparse_change_state):
- * gst/wavparse/gstwavparse.h:
- fix parsing of WAV files with non-standard fmt-tag size and fix
- skipping of unrecognized chunks... Someone please fix this thing
- to use rifflib so all this is automated.
- * sys/v4l/Makefile.am:
- * sys/v4l2/Makefile.am:
- Add X_CFLAGS because we depend on X (for overlay).
-
-2004-01-14 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c:
- Don't issue a timestamp unless we tagged the frame
- with a PTS.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_tick_callback):
- Query the audio element to get the time, not the clock. We're
- interested in the element's time here.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/aalib/gstaasink.c: (gst_aasink_chain):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- * ext/esd/esdsink.c: (gst_esdsink_chain):
- * ext/libcaca/gstcacasink.c: (gst_cacasink_chain):
- * ext/mas/massink.c: (gst_massink_chain):
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_chain):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_index),
- (gst_matroska_demux_parse_metadata):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop),
- (gst_mpeg_parse_release_locks):
- * gst/tcp/gsttcpsink.c: (gst_tcpsink_chain):
- * gst/udp/gstudpsink.c: (gst_udpsink_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_chain),
- (gst_xvideosink_release_locks):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
- use element time.
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_start),
- (gst_alsa_clock_stop):
- * gst-libs/gst/audio/audioclock.c: (gst_audio_clock_set_active),
- (gst_audio_clock_get_internal_time):
- simplify for use with new clocking code.
- * testsuite/alsa/Makefile.am:
- * testsuite/alsa/sinesrc.c: (sinesrc_init), (sinesrc_force_caps):
- fix testsuite for new caps system
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_update_metadata):
- * ext/flac/gstflacenc.c: (add_one_tag):
- length is already host endian, no need to convert. Fixes playback of
- tagged files on PPC. (bug #128384)
-
-2004-01-13 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalance.h: Adding a type to the
- colorbalance interface stating if it is hardware based or software
- based.
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_planar411):
- Removing a trailing comma.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_colorbalance_init): Integrating a patch from Jon
- Trowbridge <trow@ximian.com> querying Xv adaptor for min/max value as
- the documentation seems to be wrong on the -1000 to 1000 interval.
-
-2004-01-12 David Schleef <ds@schleef.org>
-
- * gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
- (gst_efence_buffer_alloc), (gst_fenced_buffer_new),
- (gst_fenced_buffer_default_free), (gst_fenced_buffer_default_copy):
- Fix negotiation. Add a bufferalloc function for the sink pad,
- and generally clean up some of the code.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalancechannel.c:
- (gst_color_balance_channel_dispose): Adding safety check in dispose
- method.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_interface_supported),
- (gst_xvimagesink_colorbalance_list_channels),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_colorbalance_init), (gst_xvimagesink_get_type):
- Adding colorbalance interface support to set XV parameters such as
- HUE, BRIGHTNESS, CONTRAST, SATURATION.
- * sys/xvimage/xvimagesink.h: Adding the channels list for colorbalance
- interface.
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_type),
- (gst_audio_convert_class_init), (gst_audioconvert_getcaps),
- (gst_audio_convert_init), (gst_audio_convert_set_property),
- (gst_audio_convert_get_property), (gst_audio_convert_chain),
- (gst_audio_convert_link),
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format), (plugin_init):
- - implement _getcaps and use it
- - improve linking
- - remove float caps since no float conversion is actually done
- - remove properties and arguments that were to be used for rate
- conversion
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/audio.c: (_gst_audio_structure_set_list),
- (gst_audio_structure_set_int):
- * gst-libs/gst/audio/audio.h:
- add helper functions for _getcaps matching the standard audio
- templates
-
-2004-01-12 David Schleef <ds@schleef.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Test that pad is negotiated before getting its caps.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_get_sink_element): When
- analyzing the pads of an element the bin is mostly in READY state so
- no caps were negotiated. This helper function needs to work with
- _get_caps directly then. I was not freeing them though, added that to
- fix the mem leak.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): Fixing the
- direct put buffers detection. I prefer checking GST_BUFFER_PRIVATE
- than the free_func.
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- * sys/oss/gstosselement.h:
- make an oss debugging category
- make failure more descriptive
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpeg.c:
- * ext/ffmpeg/gstffmpegcodecmap.c:
- * ext/ffmpeg/gstffmpegdec.c:
- * ext/ffmpeg/gstffmpegenc.c:
- * ext/ffmpeg/gstffmpegprotocol.c:
- * ext/gdk_pixbuf/gstgdkanimation.c:
- * ext/jpeg/gstjpeg.c:
- * ext/libpng/gstpng.c:
- * ext/mpeg2dec/perftest.c:
- * ext/speex/gstspeex.c:
- * gst-libs/gst/resample/dtos.c:
- * gst/intfloat/gstintfloatconvert.c:
- * gst/oneton/gstoneton.c:
- * gst/rtjpeg/RTjpeg.c:
- * gst/rtp/gstrtp.c:
- * sys/dxr3/dxr3init.c:
- * sys/glsink/gstgl_nvimage.c:
- * sys/glsink/gstgl_pdrimage.c:
- * sys/glsink/gstglsink.c:
- * testsuite/gst-lint:
- Make sure everybody wraps #include "config.h" in #ifdef HAVE_CONFIG_H
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_set_caps):
- * ext/faac/gstfaac.c: (gst_faac_sinkconnect):
- * ext/gdk_pixbuf/gstgdkanimation.c:
- (gst_gdk_animation_iter_create_pixbuf):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- * ext/jack/gstjack.c: (gst_jack_change_state):
- * ext/xvid/gstxviddec.c: (gst_xviddec_sink_link):
- * gst-libs/gst/play/gstplay.c: (gst_play_get_sink_element):
- * gst-libs/gst/play/play.c: (gst_play_get_sink_element):
- * gst/videofilter/gstvideofilter.c:
- (gst_videofilter_set_output_size):
- Remove all usage of gst_pad_get_caps(), and replace it with
- gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
-
-2004-01-11 David Schleef <ds@schleef.org>
+ gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (new_pad):
+ Fix non-C89 variable declaration not at the start of a block. Should
+ help some compilers.
- * configure.ac:
- * ext/Makefile.am: Fixes to make ext/libcaca compile.
- * ext/divx/gstdivxdec.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_init), (gst_gdk_pixbuf_chain): Make gdkpixbufdec
- handle images that span multiple buffers. Now work with both
- filesrc ! gdkpixbufdec and qtdemux ! gdkpixbufdec.
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- * ext/libcaca/gstcacasink.h: Fixes needed due to recent
- video/video.h changes
- * ext/xvid/gstxvid.c: (gst_xvid_csp_to_caps): same
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free): Use buffer free function instead
- of GstData free function.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_buffer_free):
- same.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_class_init),
- (gst_v4l2element_dispose), (gst_v4l2element_set_property),
- (gst_v4l2element_get_property):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_set_defaults), (gst_v4l2_open):
- add norm, channel and frequency properties.
- * sys/v4l2/gstv4l2tuner.c:
- fixes for tuner interface changes
- * sys/v4l2/gstv4l2element.h:
- * sys/v4l2/gstv4l2src.c:
- * sys/v4l2/gstv4l2src.h:
- * sys/v4l2/v4l2src_calls.c:
- * sys/v4l2/v4l2src_calls.h:
- rework v4l2src to work with saa1734 cards and allow mmaped buffers.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/tuner/tuner.c: (gst_tuner_class_init),
- (gst_tuner_find_norm_by_name), (gst_v4l2_find_channel_by_name),
- (gst_tuner_channel_changed), (gst_tuner_norm_changed),
- (gst_tuner_frequency_changed), (gst_tuner_signal_changed):
- * gst-libs/gst/tuner/tuner.h:
- GObjects aren't const.
- Add find_by_name functions.
- Add checks to _changed functions.
- * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_get_channel),
- (gst_v4l_tuner_get_norm):
- Fixes for above.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/video/video.h:
- Fix caps template names to be understandable.
- Prefix everything with GST_VIDEO.
- * ext/aalib/gstaasink.c:
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.c:
- * ext/hermes/gstcolorspace.c: (gst_colorspace_base_init):
- * ext/jpeg/gstjpegdec.c: (raw_caps_factory):
- * ext/jpeg/gstjpegenc.c: (raw_caps_factory):
- * ext/libcaca/gstcacasink.c:
- * ext/libpng/gstpngenc.c: (raw_caps_factory):
- * ext/snapshot/gstsnapshot.c:
- * ext/swfdec/gstswfdec.c:
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- * gst/chart/gstchart.c:
- * gst/deinterlace/gstdeinterlace.c:
- * gst/effectv/gsteffectv.c:
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c:
- * gst/median/gstmedian.c:
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
- (gst_monoscope_srcconnect), (gst_monoscope_chain):
- * gst/overlay/gstoverlay.c:
- * gst/smooth/gstsmooth.c:
- * gst/smpte/gstsmpte.c:
- * gst/synaesthesia/gstsynaesthesia.c:
- * gst/videocrop/gstvideocrop.c:
- * gst/videodrop/gstvideodrop.c:
- * gst/y4m/gsty4mencode.c:
- * sys/qcam/gstqcamsrc.c:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
- Make them work with new video.h file.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
- Make it work with new buffer allocation system.
-
-2004-01-11 Julien MOUTTE <julien@moutte.net>
-
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * ext/hermes/gstcolorspace.c: (gst_colorspace_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_free),
- (gst_ximagesink_buffer_alloc): Now only use GST_BUFFER_PRIVATE to keep
- a reference to everything we need.
- * sys/ximage/ximagesink.h: adding a reference to the sink in the image.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc): Now only use GST_BUFFER_PRIVATE to keep
- a reference to everything we need.
- * sys/xvimage/xvimagesink.h: adding a reference to the sink in the image
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/divx/gstdivxenc.c: remove bogus gst_caps_is_fixed() test
- * gst/debug/efence.c: (gst_efence_chain), (gst_fenced_buffer_new),
- (gst_fenced_buffer_default_copy): Fix for rename of buffer private
- structure members.
- * gst/effectv/gstwarp.c: (gst_warptv_setup): Don't reset the time
- value during a resize/renegotiation.
- * gst/videofilter/gstvideofilter.c: (gst_videofilter_chain): use
- gst_pad_alloc_buffer();
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free): Fix for rename of buffer private
- structure members.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_buffer_free):
- Fix for rename of buffer private structure members.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
- Fix for rename of buffer private structure members.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
- Fix for rename of buffer private structure members.
-
-2004-01-11 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- reviewed by: David Schleef <ds@schleef.org>
-
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstgamma.c: Gamma correction filter. Modified
- from the patch by ds to fit in with recent make_filter changes.
-
-2004-01-11 Julien MOUTTE <julien@moutte.net>
-
- * configure.ac: Adding examples/switch/Makefile
- * examples/Makefile.am: Adding examples/switch
- * examples/switch/Makefile.am: Adding switcher example.
- * examples/switch/switcher.c: (got_eos), (idle_iterate),
- (switch_timer), (main): Adding an example demonstrating switch usage
- with 2 videotestsrc showing different patterns.
- * gst/switch/gstswitch.c: (gst_switch_request_new_pad),
- (gst_switch_init): Fixing switch with the new caps system.
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ And now fix 'make distcheck' (builddir != srcdir)
- * gst-libs/gst/video/video.h:
- Fix 32bit caps. Issue remaining: The macro names are chosen poorly.
- They should probably be like
- GST_VIDEO_PAD_TEMPLATE_CAPS_{RGB,BGR,RGBx,BGRx}.
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse_trak):
- fix audio chunk size/timestamp calculation
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps):
- fix SVQ3 caps
-
-2004-01-11 Steve Baker <steve@stevebaker.org>
-
- * gst/effectv/gstaging.c: (gst_agingtv_get_type),
- (gst_agingtv_base_init), (gst_agingtv_class_init),
- (gst_agingtv_init), (gst_agingtv_setup), (gst_agingtv_rgb32),
- (gst_agingtv_set_property), (gst_agingtv_get_property):
- Port agingTV to videofilter
-
-2004-01-09 Julien MOUTTE <julien@moutte.net>
-
- * ext/hermes/gstcolorspace.c: (gst_colorspace_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
-
-2004-01-09 Julien MOUTTE <julien@moutte.net>
-
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * gst-libs/gst/xoverlay/xoverlay.c:
- (gst_x_overlay_got_desired_size): Updating doc for the xid being 0.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_init): Implementing
- the bufferalloc_function to replace bufferpools, fixing the XOverlay
- interface implementation to handle xid being 0 and fix some bugs
- triggered by Benjamin's testcase.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_init): Implementing
- the bufferalloc_function to replace bufferpools, fixing the XOverlay
- interface implementation to handle xid being 0 and fix some bugs
- triggered by Benjamin's testcase.
-
-2004-01-09 David Schleef <ds@schleef.org>
-
- * ext/librfb/gstrfbsrc.c: Hacking. Added actual decoding and
- mouse pointer events. It works.
-
-2004-01-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxenc.c: (gst_divxenc_init):
- Use explicit caps - fix capsnego.
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- Remove macro-inside-macro which caused compile errors.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_header):
- Error out if it's not a RIFF file. Else we error out without
- gst_element_error() which is not good...
-
-2004-01-08 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect):
- Fix pad_link function to handle formats that ffmpeg returns
- as multiple caps structures.
- * gst/videofilter/gstvideofilter.c: (gst_videofilter_chain):
- Only complain if source buffer is _smaller_ than expected.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
- (gst_videoscale_handle_src_event): Resize navigation events
- when passing them upstream.
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Rewrite many of the buffer painting functions to handle odd
- sizes (for many formats, size%4!=0 or size%8!=0). Most have
- been verified to work with my video card.
- * testsuite/gst-lint: Add check for elements calling
- gst_pad_get_caps() instead of gst_pad_get_allowed_caps().
-
-2004-01-08 David Schleef <ds@schleef.org>
-
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
- (gst_videodrop_link), (gst_videodrop_init): Fix negotiation.
-
-2004-01-08 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents): A
- configure event is not emiting the desired size signal. That fixes
- aspect ratio issues with gst-player.
-
-2004-01-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/median/gstmedian.c: (gst_median_link), (gst_median_init):
- Fix capsnego.
-
-2004-01-08 Julien MOUTTE <julien@moutte.net>
-
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create): Using XOverlay
- public method to fire size signal.
-
-2004-01-07 Julien MOUTTE <julien@moutte.net>
-
- * examples/gstplay/Makefile.am: Adding the interface library.
- * gst-libs/gst/play/Makefile.am: Adding the interface library.
- * gst-libs/gst/play/gstplay.c: (gst_play_set_video_sink): Connecting to the XOverlay size signal instead of GstVideoSink.
- * gst-libs/gst/play/gstplay.h: Including the XOverlay interface to check
- GST_IS_X_OVERLAY before signal connect.
- * gst-libs/gst/video/gstvideosink.c: (gst_videosink_class_init):
- Removing the have_video_size signal.
- * gst-libs/gst/video/gstvideosink.h: Removing the have_video_size signal
- and associated public method.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_sinkconnect): Using XOverlay public method to fire size
- signal.
- * sys/xvideo/xvideosink.c: (gst_xvideosink_sinkconnect),
- (gst_xvideosink_xwindow_new): Using XOverlay public method to fire size
- signal.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sinkconnect):
- Using XOverlay public method to fire size signal.
-
-2004-01-07 David Schleef <ds@schleef.org>
-
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstvideotemplate.c:
- * gst/videofilter/make_filter:
- Create gstvideoexample.c in a srcdir!=builddir friendly way.
- Convert make_filter to /bin/sh script.
-
-2004-01-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/modplug/gstmodplug.cc: fix element description
-
-2004-01-07 Julien MOUTTE <julien@moutte.net>
-
- * examples/gstplay/player.c: (got_time_tick), (got_stream_length),
- (got_video_size): Adding some new lines in g_print calls.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy), (gst_ximagesink_xwindow_resize),
- (gst_ximagesink_handle_xevents), (gst_ximagesink_fixate),
- (gst_ximagesink_sinkconnect), (gst_ximagesink_change_state),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_new),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_get_desired_size):
- Complete code review, reverting some stuff i disagree with, adding
- some fixes : time synchronization on invalid timestamps, renegotiation
- of private window.
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_destroy),
- (gst_xvimagesink_xwindow_resize), (gst_xvimagesink_handle_xevents),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_fixate), (gst_xvimagesink_sinkconnect),
- (gst_xvimagesink_change_state), (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_new),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_get_desired_size),
- (gst_xvimagesink_xoverlay_init): Complete code review, reverting some
- stuff i disagree with, adding some fixes : Renegotiation of private
- window, implementing get_desired_size.
-
-2004-01-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/audiofile/gstafsink.c: (gst_afsink_init), (gst_afsink_chain),
- (gst_afsink_handle_event):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
- * gst/avi/gstavimux.c: (gst_avimux_request_new_pad):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_init):
- Fix for instantiate-test (see core). Also remove dead code from
- jpegenc (which still needs fixing, but that's lower on my TODO
- list...).
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Never return NULL as caps.
-
-2004-01-07 David Schleef <ds@schleef.org>
+2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/cdparanoia/Makefile.am:
+ * ext/cdparanoia/gstcdparanoia.c:
+ * ext/cdparanoia/gstcdparanoia.h:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
+ (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
+ (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
+ (gst_cd_paranoia_paranoia_callback),
+ (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
+ (gst_cd_paranoia_src_set_property),
+ (gst_cd_paranoia_src_get_property), (plugin_init):
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
+ plugin again (there are still fixes required to playbin to make
+ cdda:// uris work there).
- * configure.ac:
- * ext/Makefile.am:
- * ext/librfb/Makefile.am:
- * ext/librfb/gstrfbsrc.c:
- New source plugin based on librfb-0.1. RFB (remote framebuffer)
- is the protocol used by VNC.
-
-2004-01-07 David Schleef <ds@schleef.org>
-
- * gst/videofilter/gstvideotemplate.c:
- * gst/videofilter/gstvideotemplate.h:
- * gst/videofilter/make_filter:
- Merge videotemplate header into source file.
- * gst/effectv/Makefile.am:
- * gst/effectv/gsteffectv.c: (plugin_init):
- * gst/effectv/gstwarp.c:
- Make warpTV a subclass of videofilter.
-
-2004-01-07 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- Add guard against invalid utf-8 conversions in mad. Just in case.
-
-2004-01-07 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/oss/gstosssink.c: (gst_osssink_sink_fixate):
- Fix for bug shown by poisoning
-
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps),
- (gst_v4lsrc_srcconnect), (gst_v4lsrc_getcaps), (gst_v4lsrc_get),
- (gst_v4lsrc_buffer_free):
- Fix for removed bufferpools.
-
-2004-01-07 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- Fix caps negotiation.
-
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init),
- (dvdnavsrc_update_buttoninfo), (dvdnavsrc_get),
- (dvdnavsrc_get_event_mask), (dvdnav_handle_navigation_event),
- (dvdnavsrc_event):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * gst-libs/gst/navigation/navigation.c:
- (gst_navigation_send_key_event), (gst_navigation_send_mouse_event):
- * gst-libs/gst/navigation/navigation.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_handle_src_event):
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Super-simple first version of mouse and keyboard events. Clicking
- on a DVD menu now works, although it may not take you where you
- expected.
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate):
- These fixate functions were broken - they never actually
- fixated :)
+2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Fix test case compilation.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix test case compilation.
- * ext/shout/gstshout.c: (gst_icecastsend_base_init),
- (gst_icecastsend_init):
- fix for new caps system.
- * gst-libs/gst/mixer/mixertrack.h:
- * sys/oss/gstossmixer.c: (gst_ossmixer_build_list):
- Add 'master track' flag (for tools like ACME that only want to
- change the main volume).
+2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-07 Jan Schmidt <thaytan@mad.scientist.com>
+ gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_update_duration),
+ (gst_cdda_base_src_calculate_cddb_id):
+ An integer is not a string. Fix access to uninitialised variable.
+ * tests/check/Makefile.am:
+ Add cddabasesrc unit test; also actually enable the vorbis test.
+ * tests/check/generic/states.c:
+ Blacklist new cd audio elements as well.
+ * tests/check/libs/cddabasesrc.c:
+ Unit test for GstCddaBaseSrc (discid calculation mostly).
- * ext/xvid/gstxvid.c: (gst_xvid_structure_to_csp),
- (gst_xvid_csp_to_caps):
- * ext/xvid/gstxviddec.c: (gst_xviddec_src_getcaps):
- * ext/xvid/gstxvidenc.c:
- ifdef out ARGB type when it isn't available
- in xvidcore 1.0.0beta2
+2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ Add docs for libgstcdda/GstCddaBaseSrc.
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Do one struct member per line with a semicolon at the end, that way
+ even gtk-doc might parse it without complaining.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- When we have received a new SCR right in the first buffer after
- a seek (so in the same cycle that handles the discont), we should
- handle the buffer instead of unreffing it, else we lose data.
+2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Iain <iain@prettypeople.org>
+ Add new libgstcdda with GstCddaBaseSrc class.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/base64.c:
+ * gst-libs/gst/cdda/base64.h:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
+ (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
+ (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
+ (gst_cdda_base_src_get_property),
+ (gst_cdda_base_src_get_track_from_sector),
+ (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
+ (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
+ (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
+ (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
+ (gst_cdda_base_src_uri_get_protocols),
+ (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
+ (gst_cdda_base_src_uri_handler_init),
+ (gst_cdda_base_src_setup_interfaces),
+ (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
+ (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
+ (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
+ (gst_cdda_base_src_add_tags),
+ (gst_cdda_base_src_add_index_associations),
+ (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
+ (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
+ (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
+ (gst_cdda_base_src_create):
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ * gst-libs/gst/cdda/sha1.c:
+ * gst-libs/gst/cdda/sha1.h:
+ Add new libgstcdda with GstCddaBaseSrc class.
- * gst/intfloat/gstint2float.c (gst_int2float_link): Set the
- buffer-frames caps too.
+2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/oneton/gstoneton.c (gst_oneton_sink_connect): Only create the new
- caps that we need, don't destroy them all and rebuild them. And when
- creating src pads, use the src pad template rather than the sink...
+ ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Use GstBaseSinkClass as parent_class member for class struct, not
+ GstBaseSink.
-2004-01-05 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- Add pad to element *after* setting functions such as event handler.
- Without this, the scheduler (opt) will link pads, set the event
- handler from the default event function (dispatcher in gstpad.c)
- and *after* that, we will set our own event function, which will
- thus never be used (and thus mpegdemux doesn't handle events).
+ gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_start):
+ Add start method to reset running time and number of frames sent
+ when starting up (fixes #324696; patch by: Michal Benes).
+
+2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ Add docs stuff for gnomevfssrc and gnomevfssink.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Fix example pipeline in gtk-doc blurb.
-2004-01-04 David Schleef <ds@schleef.org>
+2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
+ Original commit message from CVS:
+ * ext/gnomevfs/Makefile.am:
+ * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
+ (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
+ (gst_gnome_vfs_handle_get_type), (plugin_init):
+ * ext/gnomevfs/gstgnomevfs.h:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
+ (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
+ (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
+ (gst_gnome_vfs_sink_set_property),
+ (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
+ (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
+ (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
+ (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
+ (gst_gnome_vfs_sink_uri_get_type),
+ (gst_gnome_vfs_sink_uri_get_protocols),
+ (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
+ (gst_gnome_vfs_sink_uri_handler_init):
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Port gnomevfssink; add gtk-doc blurb.
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
+ (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_uri_get_type),
+ (gst_gnome_vfs_src_uri_get_protocols),
+ (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
+ (gst_gnome_vfs_src_uri_handler_init),
+ (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
+ (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
+ (gst_gnome_vfs_src_send_additional_headers_callback),
+ (gst_gnome_vfs_src_received_headers_callback),
+ (gst_gnome_vfs_src_push_callbacks),
+ (gst_gnome_vfs_src_pop_callbacks),
+ (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
+ (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
+ (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
+ file; add gtk-doc blurb with example pipelines.
- Fix the fixate functions to handle new prototype:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate):
- * sys/oss/gstosssink.c: (gst_osssink_sink_fixate):
- * sys/ximage/ximagesink.c: (gst_ximagesink_fixate):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
+2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * sys/ximage/ximagesink.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy), (gst_ximagesink_sinkconnect),
- (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_xoverlay_init):
- assorted fixes to make (re)embedding work
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect),
- (gst_ximagesink_get_desired_size):
- implement desired size additions to XOverlay
+=== release 0.10.1 ===
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_base_init),
- (gst_x_overlay_got_xwindow_id), (gst_x_overlay_get_desired_size),
- (gst_x_overlay_got_desired_size):
- * gst-libs/gst/xoverlay/xoverlay.h:
- Add optional "desired size" signal and querying.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.1
+ Original commit message from CVS:
+ releasing 0.10.1
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix EBML-laced block parsing. Diffs are relative to previous
- lace, not the first lace. Thanks to Mosu from the Matroska
- team for detecting this.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt), (gst_wavparse_getcaps),
- (gst_wavparse_handle_sink_event), (gst_wavparse_loop),
- (gst_wavparse_change_state):
- * gst/wavparse/gstwavparse.h:
- Quickfix for capsnego.
+ * ChangeLog:
+ * gst/typefind/gsttypefindfunctions.c:
+ iLBC30 and iLBC20 added to typefind.
+ Original commit message from CVS:
+ iLBC30 and iLBC20 added to typefind.
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/wavenc/gstwavenc.c: (set_property), (gst_wavenc_init):
- Fix indenting, fix pad creation.
+ * ChangeLog:
+ * configure.ac:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ prereleasing
+ Original commit message from CVS:
+ prereleasing
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/xvid/gstxviddec.c: (gst_xviddec_init),
- (gst_xviddec_src_getcaps), (gst_xviddec_src_link),
- (gst_xviddec_sink_link):
- Implement src_getcaps() so proper size caps is negotiated.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ stop making fun of older compilers
+ Original commit message from CVS:
+ stop making fun of older compilers
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/flac/gstflacdec.c: (gst_flacdec_loop):
- Finish flac decoder on EOS. See #116178.
+ gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ update strings, values are in microseconds
+ change the default sink buffer time to something that is smaller
+ (to help software volume mixing have a slightly lower delay) but
+ still be acceptable on Wim's laptop
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
+ Made a quack, forgot to add DUCK to the riff video template.
+
+2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
+ (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_chain):
+ Make sure pads are initialized correctly.
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add a whole bunch of FOURCC <=> MimeType.
+ Extend the riff video pad template to support the newly added fourcc.
+
+2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_activate_chain):
+ Extra debug output when activating/deactivating chains.
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
+ (is_demuxer_element), (try_to_link_1), (remove_element_chain),
+ (unlinked):
+ Remove a queue from our list when it becomes unlinked.
+ Don't add queues to elements in class 'Demux' if they
+ can only produce one pad
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_src_getcaps),
- (gst_matroska_demux_add_stream):
- * gst/matroska/matroska-ids.h:
- Add getcaps() function to fix capsnego...
+2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/video/gstvideosink.c: Add a debug category.
+ Original commit message from CVS:
+ 2005-12-18 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
+ (gst_video_sink_get_type): Add a debug category.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (gst_mpeg_demux_parse_packet), (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
- Fix more integer overflows. Again, see #126967.
+2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
+ Original commit message from CVS:
+ 2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
+ Handle downstream newsegment by sending our own newsegment before the
+ next buffer to be released. (#323900)
- * ext/mpeg2dec/gstmpeg2dec.c:
- Add support for mpeg2dec-0.4.0 (released two weeks ago). See
- #130416.
+2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
+ Original commit message from CVS:
+ 2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ add queue delay to new segment as well (as opposed to just the first
+ buffer). (bug #322347)
- * configure.ac:
- * ext/xvid/gstxvid.c: (gst_xvid_init), (gst_xvid_error),
- (gst_xvid_structure_to_csp), (gst_xvid_csp_to_caps):
- * ext/xvid/gstxvid.h:
- * ext/xvid/gstxviddec.c: (gst_xviddec_class_init),
- (gst_xviddec_init), (gst_xviddec_setup), (gst_xviddec_chain),
- (gst_xviddec_src_link), (gst_xviddec_sink_link),
- (gst_xviddec_change_state):
- * ext/xvid/gstxviddec.h:
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_profile_get_type),
- (gst_xvidenc_base_init), (gst_xvidenc_class_init),
- (gst_xvidenc_init), (gst_xvidenc_setup), (gst_xvidenc_chain),
- (gst_xvidenc_link), (gst_xvidenc_set_property),
- (gst_xvidenc_get_property), (gst_xvidenc_change_state):
- * ext/xvid/gstxvidenc.h:
- Update xvid plugin to latest xvid (1.0.0-beta3) API.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/rtp/rtp-packet.c:
- Add sys/types.h include, since OS X doesn't define in_addr_t
- in netinet/in.h, like it does on Linux (see #129600).
-
-2004-01-03 Thomas Canty <tommydal@optushome.com.au>
-
- reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_dispose):
- Correct logic of dispose function (see #129306).
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
- (gst_mpeg_parse_init):
- * gst/mpegstream/gstmpegparse.h:
- Remove clock (which was never provided, i.e. dead code), and
- also fix integer overflows at high PTS values (see #126967).
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/flac/gstflacdec.c:
- * ext/libpng/gstpngenc.h:
- * ext/mikmod/gstmikmod.h:
- OS X fixes (see #126628).
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_pad_factory),
- (gst_alsa_src_base_init): Remove bogus "src" request pad.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_base_init),
- (gst_mpeg_parse_class_init): Move pad template registration
- to class_init, since the derived class (mpegdemux) doesn't
- want them.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/ximage/Makefile.am:
- * sys/xvideo/Makefile.am:
- * sys/xvimage/Makefile.am:
- Move interface libs from LDFLAGS to LIBADD, fix relocation errors
- after installation (see #127664).
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_init),
- (gst_ffmpegenc_connect): Negotiation fixes.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_negotiate_format):
- Remove inappropriate gst_caps_free().
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Reenable Ronald's internal resize code, since the core handles
- it correctly now.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init), (gst_v4lsrc_init):
- Fix pad template stuff.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/ebml-read.c: (gst_ebml_read_sint):
- * gst/matroska/ebml-write.c: (gst_ebml_write_sint):
- fix signed integer reading/writing.
-
-2004-01-02 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/README:
- Remove outdated document
-
-2004-01-03 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/cutter/gstcutter.c: (gst_cutter_init):
- src pad was being created twice - oops.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Comment out internal resize. It doesn't handle the resulting
- XEvent internally, does another try_set_caps() which leads to
- a really nice loop.
- Real fix will come when Julien and Dave are awake. ;).
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mpeg2enc/gstmpeg2enc.cc:
- fix const/nonconst compile issue.
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate),
- (gst_xvimagesink_sinkconnect), (gst_xvimagesink_init):
- Add fixate function and a check for bad formats.
-
-2004-01-01 David Schleef <ds@schleef.org>
-
- Negotiation fixes:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
- (gst_audiofilter_init):
- * gst/debug/efence.c: (gst_efence_init):
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_link),
- (gst_deinterlace_init):
- * gst/volume/gstvolume.c: (volume_connect):
-
-2004-01-01 David Schleef <ds@schleef.org>
-
- Convert elements to use gst_pad_use_explicit_caps() where
- appropriate:
- * ext/a52dec/gsta52dec.c: (gst_a52dec_init), (gst_a52dec_reneg):
- * ext/audiofile/gstafparse.c: (gst_afparse_init),
- (gst_afparse_open_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_init),
- (gst_afsrc_open_file):
- * ext/esd/esdmon.c: (gst_esdmon_init), (gst_esdmon_get):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_init),
- (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_init), (gst_flacdec_write):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init),
- (gst_gdk_pixbuf_chain):
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_init), (gst_jpegdec_link),
- (gst_jpegdec_chain):
- * ext/mad/gstmad.c: (gst_mad_init), (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init),
- (gst_mikmod_negotiate):
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_init),
- (gst_mpeg2dec_negotiate_format):
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- * ext/speex/gstspeexdec.c: (gst_speexdec_init),
- (gst_speexdec_sinkconnect):
- * ext/swfdec/gstswfdec.c: (gst_swfdec_loop), (gst_swfdec_init):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_init),
- (gst_vorbisfile_new_link):
- * gst/ac3parse/gstac3parse.c: (gst_ac3parse_init),
- (gst_ac3parse_chain):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_audio_stream),
- (gst_asf_demux_setup_pad):
- * gst/auparse/gstauparse.c: (gst_auparse_init),
- (gst_auparse_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
- * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
- (mp1videoparse_parse_seq):
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_init),
- (bpf_from_header):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (gst_mpeg_demux_parse_pes), (gst_mpeg_demux_lpcm_set_caps):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
- (gst_mpeg_parse_send_data):
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (gst_qtdemux_add_stream):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt):
-
-2004-01-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ ext/libvisual/visual.c: change some char* into char[]
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (make_valid_name):
+ change some char* into char[]
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
+ (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ prepare to handle EOS and SEGMENT_DONE
- * configure.ac:
- Fix configure check for mpeg2enc. We need 1.6.1.93 instead of
- 1.6.1.92, since the pkg-config file of 1.6.1.92 is borked and
- it therefore uses the wrong include paths. Too bad... Note
- that 1.6.1.93 is not release yet. ;).
- Also add a check for mplex, which is now using the lib'ified
- mplex from mjpegtools, too.
- * ext/ffmpeg/gstffmpegcodecmap.c:
- Add codec_tag for 3ivx/xvid. For xvid, this should fix playback
- issues. I don't think ffmpeg handles 3ivx correctly, so this
- probably won't work. But it won't hurt either.
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_connect),
- (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect),
- (gst_ffmpegenc_chain_audio):
- Fix memleak in audio encoding. Close codec if open fails, this
- calls the cleanup routines so we can re-use the context.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Fix pad template names/types, fix memory issue with getcaps().
- * ext/mpeg2enc/gstmpeg2encoder.cc:
- * ext/mpeg2enc/gstmpeg2encoder.hh:
- Fix compile issue with new caps system (const thingy).
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.hh:
- We read a first frame right on initing, so that we have a caps
- when we init the output. This caps is cached in padprivate and
- read as first frame.
- * ext/mplex/Makefile.am:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplex.h:
- * ext/mplex/gstmplex.hh:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/mplex/gstmplexibitstream.hh:
- * ext/mplex/gstmplexjob.cc:
- * ext/mplex/gstmplexjob.hh:
- * ext/mplex/gstmplexoutputstream.cc:
- * ext/mplex/gstmplexoutputstream.hh:
- We wrap mjpegtools mplex. So I rewrote the plugin. The old plugin
- had issues, didn't do capsnego, supported only a subset of the
- mplex features and required a mplex fork in our local CVS. Plus
- that it worked agaist a very old mplex version. Rewriting was
- faster than updating it.
- * gst-libs/ext/Makefile.am:
- * gst-libs/ext/mplex/INSTRUCT:
- * gst-libs/ext/mplex/Makefile.am:
- * gst-libs/ext/mplex/README:
- * gst-libs/ext/mplex/TODO:
- * gst-libs/ext/mplex/ac3strm_in.cc:
- * gst-libs/ext/mplex/audiostrm.hh:
- * gst-libs/ext/mplex/audiostrm_out.cc:
- * gst-libs/ext/mplex/aunit.hh:
- * gst-libs/ext/mplex/bits.cc:
- * gst-libs/ext/mplex/bits.hh:
- * gst-libs/ext/mplex/buffer.cc:
- * gst-libs/ext/mplex/buffer.hh:
- * gst-libs/ext/mplex/fastintfns.h:
- * gst-libs/ext/mplex/format_codes.h:
- * gst-libs/ext/mplex/inputstrm.cc:
- * gst-libs/ext/mplex/inputstrm.hh:
- * gst-libs/ext/mplex/lpcmstrm_in.cc:
- * gst-libs/ext/mplex/mjpeg_logging.cc:
- * gst-libs/ext/mplex/mjpeg_logging.h:
- * gst-libs/ext/mplex/mjpeg_types.h:
- * gst-libs/ext/mplex/mpastrm_in.cc:
- * gst-libs/ext/mplex/mpegconsts.cc:
- * gst-libs/ext/mplex/mpegconsts.h:
- * gst-libs/ext/mplex/mplexconsts.hh:
- * gst-libs/ext/mplex/multplex.cc:
- * gst-libs/ext/mplex/outputstream.hh:
- * gst-libs/ext/mplex/padstrm.cc:
- * gst-libs/ext/mplex/padstrm.hh:
- * gst-libs/ext/mplex/stillsstream.cc:
- * gst-libs/ext/mplex/stillsstream.hh:
- * gst-libs/ext/mplex/systems.cc:
- * gst-libs/ext/mplex/systems.hh:
- * gst-libs/ext/mplex/vector.cc:
- * gst-libs/ext/mplex/vector.hh:
- * gst-libs/ext/mplex/videostrm.hh:
- * gst-libs/ext/mplex/videostrm_in.cc:
- * gst-libs/ext/mplex/videostrm_out.cc:
- * gst-libs/ext/mplex/yuv4mpeg.cc:
- * gst-libs/ext/mplex/yuv4mpeg.h:
- * gst-libs/ext/mplex/yuv4mpeg_intern.h:
- * gst-libs/ext/mplex/yuv4mpeg_ratio.cc:
- We don't fork mjpegtools' mplex in our CVS anymore.
- * gst/avi/gstavidemux.c: (gst_avi_demux_src_getcaps),
- (gst_avi_demux_add_stream):
- * gst/avi/gstavidemux.h:
- Add getcaps() function for proper caps nego. This makes some
- parts of AVI playback/reading work.
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Resize window on new capsnego. This is probably wrong, but
- I'm still committing it because with current capsnego, the
- first successfull capsnego is auto-fixated, therefore rounded
- down to the lowest values in the caps. this results in a 16x16
- XWindow that is not reized when real capsnego finishes.
- Dave, I see more cases of this, do you know a proper solution?
- * tools/gst-launch-ext.in:
- Fix MPEG-4 AAC (Apple iPod/iTunes) file commandline.
-
-2003-12-31 David Schleef <ds@schleef.org>
-
- * gst/tcp/gsttcpsrc.c: (gst_tcpsrc_get):
- * gst/udp/gstudpsrc.c: (gst_udpsrc_get):
- Change gst_pad_proxy_link() to gst_pad_try_set_caps()
-
-2003-12-30 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info), (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_pad_link), (gst_ffmpegcsp_init),
- (gst_ffmpegcsp_chain): Negotiation fixes
- * ext/mad/gstmad.c: (gst_mad_chain): Negotiation fixes
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_link), (gst_audio_convert_channels):
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_getcaps),
- (gst_audioscale_link), (gst_audioscale_get_buffer),
- (gst_audioscale_chain): Negotiation fixes
- * gst/audioscale/gstaudioscale.h:
- * gst/videofilter/gstvideofilter.c:
- (gst_videofilter_format_get_structure), (gst_videofilter_getcaps),
- (gst_videofilter_link), (gst_videofilter_init),
- (gst_videofilter_set_output_size), (gst_videofilter_setup),
- (gst_videofilter_find_format_by_structure):
- * gst/videofilter/gstvideofilter.h: Negotiation fixes
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps),
- (gst_videoscale_link):
- * gst/videoscale/videoscale.c: (videoscale_get_structure),
- (videoscale_find_by_structure), (gst_videoscale_setup):
- * gst/videoscale/videoscale.h: Negotiation fixes
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_fixate), (gst_ximagesink_init): Add a fixate
- function, restrict resizing to a multiple of 4 (hack until
- everyone supports odd sizes correctly).
-
-2003-12-29 Colin Walters <walters@verbum.org>
-
- * ext/esd/esdsink.c (gst_esdsink_link): Fix typo; get depth instead of
- signed.
-
-2003-12-30 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/sndfile/gstsf.c: (gst_sf_loop):
- Fix warning about discarding const qualifier
-
-2003-12-27 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/cutter/gstcutter.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/volenv/gstvolenv.c:
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/video/video.c:
- Fix warnings
+2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ tests/check/generic/states.c: Blacklist cdparanoia element in state test.
+ Original commit message from CVS:
+ * tests/check/generic/states.c: (GST_START_TEST):
+ Blacklist cdparanoia element in state test.
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio):
- Don't send ALSA debugging to stderr.
- * ext/alsa/gstalsa.h:
- Use GST_WARNING instead of g_warning when ALSA functions fail.
+2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
+ Original commit message from CVS:
+ * gst/tcp/gsttcp.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ Add <string.h> includes for memset and FD_ZERO (fixes #323878;
+ patch by: Benjamin Pineau).
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Free XVAdapterInfo correctly.
+2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
+ (gst_video_rate_chain):
+ Fix timestamping for videorate when the first buffer it sees has a
+ non-zero timestamp. Fix some misleading debug output.
- * ext/mad/gstid3tag.c: (gst_id3_tag_add_src_pad),
- (gst_id3_tag_do_caps_nego), (gst_id3_tag_src_link):
- Make id3tag use correct caps nego.
+2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
-2003-12-27 Amaury Jacquot <sxpert@esitcom.org>
+ gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ Don't leak all input buffers to audioresample.
- * ext/ivorbis/vorbis.c:
- * ext/ivorbis/vorbisenc.h:
- * ext/ivorbis/vorbisfile.c:
- Modify so that it uses the new caps things
+2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
+ Don't operate on empty text buffers. Strip newlines and
+ tabs only from the end of the text, but leave them intact
+ in the middle. Fix typo in gtk-doc description.
- * testsuite/spider/spider1.c: (main):
- * testsuite/spider/spider2.c: (main):
- * testsuite/spider/spider3.c: (main):
- Make tests compile again. They probably don't work.
+2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-24 Colin Walters <walters@verbum.org>
+ gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin.c: (handoff):
+ Make sure the video frame buffer we return to apps via the
+ "frame" property always has caps set on it. Modify
+ _gst_gvalue_set_object() macro to handle NULL objects
+ gracefully too.
- * sys/oss/gstosssink.c (gst_osssink_sink_fixate): Return NULL if
- we can't fixate the caps anymore.
+2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_init),
+ (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
+ (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
+ (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Adjust to some recent api changes and add wtays new cool seeking
+ capabillities
+
+2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
+ Original commit message from CVS:
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsadeviceprobe.h:
+ Helper functions to add device probing via the GstPropertyProbe
+ interface to a class.
+ * ext/alsa/gstalsamixer.h:
+ Comment out GST_ALSA_MIXER, it returns a struct that's not
+ used.
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
+ Add some debug info.
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_interface_supported),
+ (gst_implements_interface_init),
+ (gst_alsa_mixer_element_init_interfaces),
+ (gst_alsa_mixer_element_class_init),
+ (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
+ (gst_alsa_mixer_element_set_property),
+ (gst_alsa_mixer_element_get_property),
+ (gst_alsa_mixer_element_change_state):
+ * ext/alsa/gstalsamixerelement.h:
+ Add 'device' and 'device-name' properties. Add GstPropertyProbe
+ for device handling (gnome-volume-control will need that).
-2003-12-23 David Schleef <ds@schleef.org>
+2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/volume/gstvolume.c: (volume_init): Proxy getcaps.
- * sys/oss/gstosssink.c: (gst_osssink_init),
- (gst_osssink_sink_fixate): Add fixate function.
+ * ChangeLog:
+ * ext/Makefile.am:
+ * gst-plugins-base.spec.in:
+ updates to activate cdparanoia plugin
+ Original commit message from CVS:
+ updates to activate cdparanoia plugin
-2003-12-24 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
- * ext/ffmpeg/gstffmpegcodecmap.c:
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_srcconnect_func), (gst_ffmpegcsp_sinkconnect),
- (gst_ffmpegcsp_srcconnect), (gst_ffmpegcsp_get_type),
- (gst_ffmpegcsp_base_init), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_chain),
- (gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
- (gst_ffmpegcsp_get_property), (gst_ffmpegcsp_register):
- fix typo in RGB masks, and move back to "old" colorspace
- capsnego code until whoever wrote this new crap has actually
- tested it so that it works.
- And yes, this works, keep it that way please.
+ ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
+ Use the correct function to free list of typefind factories.
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_init),
+ (gst_video_test_src_parse_caps), (gst_video_test_src_query),
+ (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
+ (gst_video_test_src_create):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Implement seeking in videotestsrc.
+ Small cleanups.
+
+2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
+ Original commit message from CVS:
+ * ext/cdparanoia/Makefile.am:
+ * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
+ (gst_paranoia_endian_get_type), (_do_init),
+ (cdparanoia_class_init), (cdparanoia_init),
+ (cdparanoia_set_property), (cdparanoia_get_property),
+ (cdparanoia_do_seek), (cdparanoia_is_seekable),
+ (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
+ (cdparanoia_convert), (cdparanoia_get_query_types),
+ (cdparanoia_query), (cdparanoia_set_index),
+ (cdparanoia_uri_set_uri):
+ * ext/cdparanoia/gstcdparanoia.h:
+ Partially ported cdparanoia now that basesrc can support a
+ plugin like this..
+
+2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
+ Original commit message from CVS:
+ * tests/examples/seek/scrubby.c: (main):
+ Set higher priority for bus events so they don't get reordered with
+ gtk gui events.
+ * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
+ (flush_toggle_cb), (main):
+ Added checkbox do disable flushing seeks.
+ Disable scrubbing when doing non flushing seeks.
+
+2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
+ (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
+ (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
+ (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
+ Implement some sort of event handling that doesn't rely on
+ g_return_if_fail; make sure we always push the last chunk of an
+ .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
+ state change function; remove some old cruft. Seeking is still
+ rather unlikely to work though.
+ * tools/.cvsignore:
+ Ignore more.
- * ext/divx/gstdivxdec.c: (gst_divxdec_base_init),
- (gst_divxdec_init), (gst_divxdec_negotiate):
- * ext/divx/gstdivxdec.h:
- * ext/divx/gstdivxenc.c: (gst_divxenc_base_init),
- (gst_divxenc_init):
- * ext/faac/gstfaac.c: (gst_faac_base_init), (gst_faac_init),
- (gst_faac_sinkconnect), (gst_faac_srcconnect):
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encoder.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_base_init),
- (dxr3audiosink_init), (dxr3audiosink_pcm_sinklink):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_base_init),
- (dxr3spusink_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_base_init),
- (dxr3videosink_init):
- Fix caps breakage after Dave's caps branch merge.
+2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
+ Original commit message from CVS:
+ 2005-12-11 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
+ Fixed a leak of the current image reference when cleaning up.
+ Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- Fix for 24bpp display.
+2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
-2003-12-23 Colin Walters <walters@verbum.org>
+ tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
+ Original commit message from CVS:
+ * tools/Makefile.am:
+ * tools/gst-launch-ext-m.m:
+ Remove gst-launch-ext. It doesn't work, and is no longer
+ particularly useful.
- * ext/gnomevfs/gstgnomevfssink.c: Add ARG_HANDLE property that takes
- a GnomeVFSHandle directly.
+2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
+ don't pass random values to ogmparse convert function.
+ Make seeking possible in the exile1.ogm file.
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (volume_connect), (volume_parse_caps),
- (volume_base_init), (volume_init):
- Reenable volume element and fix to work with new caps stuff.
- Rhythmbox needs this.
+2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
+ Work around refcount problem with g_value_set_object() that occur
+ if the core has been compiled against GLib-2.6 (g_value_set_object()
+ will only g_object_ref() the element, but the caller will
+ gst_object_unref() it and bad things will happen due to the way
+ GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
+ totem for people on FC4 using Thomas's 0.10 RPMs.
- * gst/qtdemux/qtdemux.c: (plugin_init):
- qtdemux requires bytestream
+2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ Time to welcome ogm to 0.10 :)
+ Original commit message from CVS:
+ Time to welcome ogm to 0.10 :)
+ * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
+ (gst_ogg_pad_typefind):
+ Oggdemux can now properly typefind elements with dynamic pads.
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
+ Properly set caps on src pad, and set caps on outgoing buffers.
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_link):
- Fix remaining caps handling errors due to CAPS merge.
+2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * ChangeLog:
+ * ext/alsa/gstalsamixer.h:
+ * ext/alsa/gstalsamixerelement.h:
+ * ext/alsa/gstalsamixeroptions.h:
+ * ext/alsa/gstalsamixertrack.h:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasink.h:
+ * ext/alsa/gstalsasrc.c:
+ * ext/alsa/gstalsasrc.h:
+ * ext/cdparanoia/gstcdparanoia.h:
+ * ext/gnomevfs/gstgnomevfsuri.h:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gsttextoverlay.h:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisenc.h:
+ * ext/vorbis/vorbisparse.h:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/colorbalancechannel.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ * gst-libs/gst/interfaces/navigation.h:
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ * gst-libs/gst/interfaces/tuner.h:
+ * gst-libs/gst/interfaces/tunerchannel.h:
+ * gst-libs/gst/interfaces/tunernorm.h:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.h:
+ * gst-libs/gst/riff/riff-read.h:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/video/video.h:
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioconvert/gstchannelmix.c:
+ * gst/audioconvert/gstchannelmix.h:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/resample.h:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreaminfo.h:
+ * gst/tcp/gstfdset.c:
+ * gst/tcp/gstfdset.h:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcp.h:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpclientsrc.h:
+ * gst/tcp/gsttcpplugin.h:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.h:
+ * sys/v4l/gstv4lcolorbalance.h:
+ * sys/v4l/gstv4ltuner.h:
+ * sys/v4l/gstv4lxoverlay.h:
+ * sys/v4l/v4l_calls.h:
+ * sys/v4l/videodev_mjpeg.h:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/audioresample.c:
+ * tests/check/elements/audiotestsrc.c:
+ * tests/check/elements/videotestsrc.c:
+ * tests/check/elements/volume.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ expand tabs
+ Original commit message from CVS:
+ expand tabs
- * ext/faad/gstfaad.c: (gst_faad_base_init), (gst_faad_init),
- (gst_faad_sinkconnect), (gst_faad_srcgetcaps),
- (gst_faad_srcconnect):
- Port to new caps system.
+2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-21 Julien MOUTTE <julien@moutte.net>
+ * docs/libs/tmpl/gstaudio.sgml:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/libs/tmpl/gstgconf.sgml:
+ * docs/libs/tmpl/gstmixer.sgml:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * docs/libs/tmpl/gsttuner.sgml:
+ * docs/libs/tmpl/gstxoverlay.sgml:
+ put back stability level
+ Original commit message from CVS:
+ put back stability level
- * examples/gstplay/player.c: (got_time_tick), (got_stream_length),
- (got_video_size), (main): using g_print instead of g_message.
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup): Fixing EOS
- signal which was not emitted because of "switch" element added to the
- bin but not connected. (Removing from the bin temporarily)
+2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-21 Julien MOUTTE <julien@moutte.net>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * configure.ac: X_DISPLAY_MISSING is set to 1 if AC_PATH_XTRA fails to
- find X development files. I don't understand the previous tests and
- they fail on my debian/ppc unstable. This one works.
- * examples/gstplay/player.c: (main): Set the pipeline to READY before
- exiting.
- * gst-libs/gst/play/gstplay.c: (gst_play_get_length_callback),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization): Add some safety checks in set_ methods
- and state_change. This was throwing some ugly CRITICAL messages when
- pipeline was getting disposed and casts were failing.
+=== release 0.10.0 ===
-2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- Improve mpeg2enc detection. This is for distributions that do
- ship mjpegtools, but without mpeg2enc. Also does object check
- for might there ever be ABI incompatibility.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Add Andrew as second maintainer (he's helping me), and also add
- an error if no caps was set. This happens if I pull before capsnego
- and that's something I should solve sometime else.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix time parsing.
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
- (gst_matroska_mux_track_header):
- Add caps to templates.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
- Add mpegversion=1 to prevent confusion with MPEG/AAC.
- * gst/mpegstream/gstmpegdemux.c:
- Remove layer since it causes warnings about unfixed caps.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- Fix obvious typo (we error out if caps were set, we should of
- course error out if *no* caps were set).
- * sys/oss/gstosselement.c: (gst_osselement_convert):
- Fix format conversion, we confused bits/bytes.
- * sys/oss/gstosselement.h:
- Improve documentation for 'bps'.
- * sys/v4l/TODO:
- Remove stuff about plugins that need removing - this was done
- ages ago.
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
- (gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
- (gst_v4lsrc_src_query):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
- (gst_v4l2src_src_convert), (gst_v4l2src_src_query):
- Add get_query_types(), get_formats() and query() functions.
-
-2003-12-21 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ChangeLog: moved to gstreamer/docs/random/old/ChangeLog.gst-plugins
- * moved CVS to freedesktop.org
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.0
+ Original commit message from CVS:
+ releasing 0.10.0
+
projects / platform / upstream / gstreamer.git / blobdiff