-2009-01-05 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.30 ===
- * 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 Alessandro Decina <alessandro.d@gmail.com>
+2010-07-15 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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 Jan Schmidt <jan.schmidt@sun.com>
-
- * win32/common/libgstaudio.def:
- * win32/common/libgsttag.def:
- * win32/common/libgstvideo.def:
- Add new exports to win32 files.
+ releasing 0.10.30, "Difficult Birth"
-2008-12-31 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-07-15 00:32:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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 Edward Hervey <edward.hervey@collabora.co.uk>
+ * po/cs.po:
+ * po/lv.po:
+ po: update translations
- * 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.
+2010-07-14 12:59:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-31 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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.
- * 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.
+2010-07-08 14:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-30 Edward Hervey <edward.hervey@collabora.co.uk>
+ * docs/design/Makefile.am:
+ docs: dist more of the gst-plugin-base design docs
- * 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.
+2010-07-07 00:35:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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
- * 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.
+2010-07-07 00:24:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * gst/playback/gstplaybin2.c: (no_more_pads_cb):
- Add some debug info.
+2010-07-06 09:47:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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>
-2008-12-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Use correct Y offset for YVYU -> RGB conversions
+ Fixes bug #623530.
- * 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.
+2010-07-04 17:26:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Use correct Y offset for the YUY2 -> RGB conversions
+ Fixes bug #623530.
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- Make GstAudioSrcSlaveMethod get_type() function non-static
- as it's public now.
+2010-07-04 14:55:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstaudio.def:
- * win32/common/libgstnetbuffer.def:
- Add some missing functions to the list of exported symbols.
+ * docs/plugins/inspect/plugin-ogg.xml:
+ docs: update ogg introspection info after riff fourcc addition
-2008-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-07-02 20:09:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Andrew Feren <acferen at yahoo dot com>
+ * 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.
- * 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.
+2010-07-02 14:56:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-17 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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).
+2010-07-02 13:59:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-16 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix invalid memory accesses with odd widths/heights during subsampling
+ Fixes bug #623375.
- * 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.
+2010-07-01 21:21:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-13 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- * win32/common/config.h:
- Update to CVS version.
+2010-06-30 21:20:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/riff/riff-media.c:
+ riff: add FLV4 fourcc and map it to video/x-vp6-flash
+ Fixes #623176.
-2008-12-13 Edward Hervey <bilboed@gmail.com>
+2010-06-30 15:13:10 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * m4/Makefile.am:
- And a couple more .m4 that don't exist anymore with gettext 0.17
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: declare with G_DEFINE_TYPE for type safety
+ Fixes #623233.
-2008-12-13 Edward Hervey <edward.hervey@collabora.co.uk>
+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.
- * 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.
+2010-06-30 11:00:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * configure.ac:
+ configure: keep things sorted alphabetically
+ On special request. Because it's important, apparently.
- * 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.
+2010-06-29 18:48:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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:
+ 0.10.29.3 pre-release
- * 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.
+2010-06-29 18:46:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-12 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ configure: fix --disable-external
- * gst/playback/gstplaybin.c:
- * gst/playback/gstplaybin2.c:
- XRef to GstXOverlay.
+2010-06-28 15:43:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+ * autogen.sh:
+ * configure.ac:
+ Bump automake requirement to 1.10
+ For maintainability reasons and $(builddir).
+ Fixes #622944.
- * 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.
+2010-06-27 10:43:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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.
- * 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.
+2010-06-26 21:28:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/videotestsrc/Makefile.am:
+ videotestsrc: Explicitely link with $(LIBM)
- * 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.
+2010-06-26 21:27:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-11 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/videoscale/Makefile.am:
+ videoscale: Explicitely link with $(LIBM)
- * 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
+2010-06-26 18:19:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.29.2 pre-release
- Patch by: Guillaume Emont <guillaume at fluendo dot com>
+2010-06-26 18:19:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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
-2008-12-11 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-26 18:19:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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
-2008-12-11 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-26 17:55:12 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- Pushing 10 buffers is enough to run the test.
+ * 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
-2008-12-11 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-26 12:39:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * docs/design/design-orc-integration.txt:
+ docs: fix a few typos
-2008-12-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-26 12:03:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybin2.c: (pad_added_cb):
- Error out with a missing-plugin error when the input-selector was not
- found.
+ * 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.
- * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
- Indentation.
+2010-06-26 11:38:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-10 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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/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.
+2010-06-25 17:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-12-10 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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.
+2010-06-24 13:30:59 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-12-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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
- Patch by: Luis Menina <liberforce at freeside dot fr>
+2010-06-24 15:31:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2008-12-09 Julien Moutte <julien@fluendo.com>
+2010-06-24 15:13:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
+ * tests/check/gst/typefindfunctions.c:
+ tests: add test that runs all typefinders over random data
-2008-12-09 Stefan Kost <ensonic@users.sf.net>
+2010-06-06 12:31:35 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst-libs/gst/riff/riff-read.c:
- Fix handling of odd chunks in riff metadata.
+ * 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
-2008-12-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-24 13:42:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * .gitignore:
+ .gitignore: add temporary orc test directory
-2008-12-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-24 13:30:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * tests/check/Makefile.am:
+ tests: add plugin loading whitelist to test environment
+ Only want to load core/-base plugins here.
-2008-12-08 Stefan Kost <ensonic@users.sf.net>
+2010-06-24 15:09:04 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstdecodebin2.c:
- Add basic docs to decodebin and link to decodebin from decodebin2.
-
-2008-12-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From 73ff93a to a519571
- Patch by: Olivier Crete <tester at tester ca>
+2010-06-24 08:41:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Implement gst_rtcp_packet_remove(). Fixes #563174.
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ tag: Fix printf format string
+ Use %s for strings, not %d.
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add unit test for some RTCP functions.
+2010-06-24 12:06:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-04 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix resampling of ARGB scanlines
+ Previously we would read behind the end of the source lines.
- * configure.ac:
- Apparently AC_CONFIG_MACRO_DIR breaks when using more
- than one macro directory, reverting last change.
+2010-06-16 14:08:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-12-04 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * configure.ac:
- Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
- our M4 macros.
+2010-06-16 11:19:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-12-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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.
- * 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
+2010-06-23 12:10:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- Patch by: Cygwin Ports maintainer
- <yselkowitz at users dot sourceforge dot net>
+2010-06-23 10:16:07 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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.
-2008-11-28 Christian Schaller <christian.schaller@collabora.co.uk>
+2010-06-18 16:37:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/speexresample/Makefile.am: fix build by removing -lgstaudio
+ * 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
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-18 14:17:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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.
+ * 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
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-14 12:27:02 +0200 Philippe Normand <pnormand@igalia.com>
- * gst/speexresample/README:
- Update README with the latest diff between the Speex resampler
- and our copy.
+ * 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
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-17 17:07:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/speexresample/gstspeexresample.c: (plugin_init):
- Update the debug category from speex_resample to audioresample.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add YUY2/YVYU to all RGB formats conversions
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-17 16:57:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y42B to YUY2/YVYU/UYVY conversion for odd widths
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-17 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/inspect/plugin-audioresample.xml:
- Regenerated for library filename change.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix YUY2/YVYU/UYVY to Y42B conversion for odd widths
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-17 16:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * common:
* 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/gst-plugins-base-plugins.signals:
* 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.
+ 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.
-2008-11-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-16 19:15:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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!
+ * 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.
-2008-11-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-15 16:49:17 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: 이문형 <iwings at gmail dot com>
+ * common:
+ Automatic update of common submodule
+ From 9339ccc to 35617c2
- * 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.
+2010-06-15 16:53:49 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From 5adb1ca to 9339ccc
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- Fix typo in the docs.
+2010-06-15 16:34:54 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From 57c89b7 to 5adb1ca
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
- If no stream was found before receiving EOS, post an error message.
- Fixes #561924.
+2010-06-15 15:32:34 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From c804988 to 57c89b7
- * 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.
+2010-06-15 13:09:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/audioresample.c:
+ Revert "audioresample: set pads as negotiable"
+ This reverts commit 5f74f3a82eb54f9a9517f99dffbe45ce4d474870.
- * gst/playback/gstplaybin2.c: (activate_group):
- Fix buffer-duration property.
+2010-06-15 13:09:29 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/audioconvert.c:
+ Revert "audioconvert: set pads negotiable"
+ This reverts commit bbd7dee8f604bd0373a82e6e5cc3eec8313806ac.
- * 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.
+2010-06-14 15:19:32 -0700 David Schleef <ds@schleef.org>
-2008-11-24 Michael Smith <msmith@songbirdnest.com>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix black horizontal line in image
- * 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.
+2010-06-14 15:05:16 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-11-24 Stefan Kost <ensonic@users.sf.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
- * 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.
+2010-06-14 18:10:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-11-24 David Schleef <ds@schleef.org>
+ * tests/check/elements/audioconvert.c:
+ audioconvert: set pads negotiable
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- If we're muxing a dirac stream, flush the page after every picture.
+2010-06-14 17:48:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-11-24 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/audioresample.c:
+ audioresample: set pads as negotiable
- * 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.
+2010-06-14 16:25:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-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.
+ * common:
+ Revert accidental downgrade of common revision.
- * 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.
+2010-06-14 16:07:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/videoscale.c:
+ videoscale: And only expect a single buffer in the unit test
- * gst/videorate/gstvideorate.c:
- Add jpeg and png image media types to the caps. Fixes #561436.
+2010-06-14 16:02:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/videoscale.c:
+ videoscale: Only convert one buffer instead of five
+ Should make the unit test a lot faster.
- * 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.
+2010-06-14 14:13:32 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-21 David Schleef <ds@schleef.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix unitialized variables
+ yay macosx compilers :(
- Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
+2010-06-14 14:13:16 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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-libs/gst/video/video.c:
+ video: Fix unitialized variable.
+ yay macosx compilers :(
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-14 13:27:01 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstdecodebin2.c:
- Fix random fat-fingering making this not compile.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use Quarks for structure name/field checking
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-14 13:26:02 +0200 Edward Hervey <bilboed@bilboed.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/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.
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-14 13:24:06 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Fix win32 build. Oops.
+ * common:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Transfer structures instead of copying them
+ Avoids many expensive structure copies
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-14 13:20:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Use WSAGetLastError() rather than errno/h_errno on win32.
+ * configure.ac:
+ configure: Use GLIB_EXTRA_CFLAGS
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-14 13:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c:
- Support WMA Lossless properly.
+ * common:
+ Automatic update of common submodule
+ From 7a0fdf5 to c804988
-2008-11-18 David Schleef <ds@schleef.org>
+2010-06-14 11:31:09 +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).
+ * common:
+ Automatic update of common submodule
+ From 6da3bab to 7a0fdf5
-2008-11-18 Alessandro Decina <alessandro.d@gmail.com>
+2010-06-14 11:20:00 +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/examples/seek/jsseek.c:
+ jsseek: Don't use deprecated GLib API
+ Fixes once again bug #605100.
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-14 11:16:45 +0200 Prahal <prahal at yahoo.com>
- * 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/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/typefind/gsttypefindfunctions.c:
- Make sure we never jump backwards when typefinding corrupt mov files.
+2010-06-14 11:11:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * gst-libs/gst/interfaces/propertyprobe.c:
- Fix random type causing a docs warning.
+2010-06-14 11:08:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-14 Stefan Kost <ensonic@users.sf.net>
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfssrc: Fix possible NULL pointer dereference
+ It's always an error if gst_buffer_try_new_and_alloc() returns NULL
- * sys/v4l/gstv4l.c:
- Give it a minimal rank for autovideosrc.
+2010-06-14 11:03:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-13 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Return FALSE from the seek handler if no seek callback was set
- * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
- (plugin_init):
- Improve typefinding of ISO JPEG2000 mime types.
+2010-06-14 09:53:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/gio/gstgiostreamsrc.c:
+ giostreamsrc: Fix copy&paste error in the docs
- * 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>
+2010-06-14 00:33:04 +0100 Tim-Philipp Müller <tim.muller@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/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'".
-2008-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-06-13 22:17:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c:
- Change author string after seeing output of gst-inspector.
+ * tests/check/elements/videoscale.c:
+ videoscale: Add ffmpegcolorspace after videotestsrc for the unit test
-2008-11-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-13 20:57:19 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: ...and add Y16 case for the linear scaling
-2008-11-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-13 20:38:23 +0200 Sebastian Dröge <sebastian.droege@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/videoscale/gstvideoscale.c:
+ videoscale: Add Y16 case for 4-tap scaling
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-13 18:27:19 +0200 Sebastian Dröge <sebastian.droege@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.
+ * tests/check/Makefile.am:
+ tests: Fix linking of the tags test
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-13 08:20:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Only convert caps to string when debug is enabled.
+ * gst-libs/gst/video/video.h:
+ video: Re-add (but deprecated) GST_VIDEO_{RED,GREEN,BLUE}_MASK_1[56]
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-12 21:04:48 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use correct variables for debug output
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-12 16:51:41 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y16 from/to GRAY8 conversion
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-12 16:31:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Based on patch by: Matthias Kretz <kretz at kde dot org>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Don't crash when doing gray YUV to GRAY conversion
- * 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
+2010-06-12 16:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ videoscale: Update disted orc files
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+2010-06-12 16:16:37 +0200 Sebastian Dröge <sebastian.droege@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/gsturidecodebin.c:
+ uridecodebin: Allow video/webm for progressive downloading
-2008-10-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-12 13:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Nick Haddad <nick at haddads dot net>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for more gray formats
- * 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.
+2010-06-01 16:45:34 +0000 Martin Bisson <martin.bisson@gmail.com>
-2008-10-30 Stefan Kost <ensonic@users.sf.net>
+ * 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/audioresample/gstaudioresample.c:
- Return the result of parent_class->event().
+2010-06-01 14:42:54 +0000 Martin Bisson <martin.bisson@gmail.com>
-2008-10-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * 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.
+2010-06-12 13:35:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Use libgstvideo for caps parsing, etc
- * win32/common/libgstaudio.def:
- Add new symbols.
+2010-06-12 13:04:43 +0200 Philippe Normand <phil@base-art.net>
-2008-10-23 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix format string compiler warning on OS X
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- Attempt to make obfuscated code clearer.
+2010-06-12 13:00:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use GST_VIDEO_CAPS_GRAY{8,16}
- * 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.
+2010-06-12 12:57:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Implement linear merging of Y16 scanlines with orc
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h:
- Don't mark as gtk-doc docs as they aren't public.
+2010-06-12 08:26:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ Automatic update of common submodule
+ From 733fca9 to 6da3bab
- * 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
+2010-06-11 22:16:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-22 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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).
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Remove useless buffer size assignment. It already has this value.
+2010-06-11 12:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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.
+2010-06-10 18:12:37 +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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add typefinder for dts audio
- * gst-libs/gst/audio/gstringbuffer.h:
- Update some docs.
+2010-06-11 15:23:14 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix unitialized variables
+ I hate thee macosx
- * 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.
+2010-06-11 08:47:27 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix debug statements
- * tests/icles/.cvsignore:
- update ignore file.
+2010-06-11 08:47:17 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/tag/gstexiftag.c:
+ exiftag: Fix unitialized variable
-2008-10-17 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-10 20:45:42 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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()
+ * win32/common/libgsttag.def:
+ win32: update def file
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-10 20:36:32 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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-libs/gst/tag/tag.h:
+ docs: fix gtk-doc warnings
+ Variable names in function prototypes in the headers should match the doc-
+ comment.
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-10 08:47:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/gstplaybin2.c:
+ playbin2: If the text-sink claims to support ANY caps assume it only support raw plaintext subtitles
+ Fixes bug #621071.
-2008-10-16 Stefan Kost <ensonic@users.sf.net>
+2010-06-10 08:46:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/icles/playbin-text.c:
+ icles: Only accept plain subtitles in the playbin-text icles test
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-09 22:34:24 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for VP6F (On2 VP6 Flash variant)
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-09 12:35:40 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * configure.ac:
+ Use the Orc m4 macro
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-09 12:40:00 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From fad145b to 733fca9
-2008-10-15 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-09 12:33:51 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 47683c1 to fad145b
-2008-10-14 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-09 15:58:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_decode_buffer):
- Fix build on macosx.
+ * 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
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-10 14:01:46 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Based on patch by: Robin Stocker <robin at nibor dot org>
+ * 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>
- * 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.
+ * 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
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-03 23:02:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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.
+ * 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-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps):
- Propagate return value of setcaps.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle SEEKING query in push mode too
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-09 16:38:34 +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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Update the total time from the Skeleton 4 indexes
+ Fixes bug #620939, see bug #607945.
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-09 16:33:25 +0200 Sebastian Dröge <sebastian.droege@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.
+ * ext/ogg/gstoggstream.c:
+ oggstream: Implement latest version of the Skeleton 4.0 spec
+ Fixes bug #620939.
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-09 16:59:10 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- For looking at the 4th byte we have to get 4 bytes of course
- and not 3.
+ * 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.
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-08 13:34:46 +0100 Tim-Philipp Müller <tim.muller@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.
+ * .gitignore:
+ .gitignore: add orc-related temp files
-2008-10-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-08 13:26:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Don't assert on caps==NULL.
+ * 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.
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-08 13:11:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Implement correct parsing of Skeleton 4.0 index packets
-2008-10-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-08 12:01:15 +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_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.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Add parsing of Skeleton 4.0 indexes
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-08 11:40:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Based on a patch by: xavierb at gmail dot com
+ * 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.
- * 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.
+2010-06-08 11:31:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Parse Skeleton stream major/minor version
- * ext/vorbis/vorbisenc.c:
- (gst_vorbis_enc_buffer_check_discontinuous):
- Fix discontinuity detection which was broken by last commit.
+2010-06-08 11:26:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-09 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Use binary search for searching in the index
- * configure.ac::
- Require core CVS for ghostpad API additions used by decodebin2.
+2010-06-08 11:02:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-08 Edward Hervey <edward.hervey@collabora.co.uk>
+ * tests/check/libs/video.c:
+ video: Fix unit test, the Y800 checks were not used before and were not working
- * 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.
+2010-06-08 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/video/video.c:
+ video: Return TRUE in gst_video_format_is_gray() for Y800 and Y16
- * 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.
+2010-06-08 00:33:31 -0700 David Schleef <ds@schleef.org>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videotestsrc/gstvideotestsrcorc-dist.c:
+ audioconvert, videotestsrc: Update generated Orc code
+ Fixes compile errors with initialization of unions.
- Patch by: Pavel Zeldin <pzeldin at gmail dot com>
+2010-06-08 00:32:36 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * REQUIREMENTS:
+ requirements: change liboil to Orc
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-06 23:50:05 -0700 David Schleef <ds@schleef.org>
- Patch by: Jan Gerber <j at oil21 dot org>
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: convert from liboil to orc
- * 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.
+2010-06-06 23:48:35 -0700 David Schleef <ds@schleef.org>
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/Makefile.am:
+ tests: Add orc tests
- Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+2010-06-06 23:48:15 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Implement skew clock slaving. Fixes #552559.
+ * 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
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:47:53 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/multichannel.c:
- * gst-libs/gst/audio/testchannels.c:
- Fix include of config.h
+ * 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
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:47:16 -0700 David Schleef <ds@schleef.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
- Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
+2010-06-06 23:46:41 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:45:58 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:45:10 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain):
- Improve debugging of the rtptime.
+ * docs/design/Makefile.am:
+ * docs/design/design-orc-integration.txt:
+ docs: Add notes about Orc integration
-2008-10-05 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-06 23:34:39 -0700 David Schleef <ds@schleef.org>
* configure.ac:
- Back to development -> 0.10.21.1
+ configure: convert liboil check to orc
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-08 07:34:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggmux.c:
+ oggmux: Start a new page for every CMML buffer
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-07 14:38:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
- (plugin_init):
- Add typefinder for MXF.
+ * 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
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-07 08:31:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/Makefile.am:
- Only build test-colorkey if GTK+ is available.
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: Map "Y8 " and "GREY" to "Y800" and add it to the template caps
-=== release 0.10.21 ===
+2010-06-07 08:17:13 +0200 Martin Bisson <martin.bisson@gmail.com>
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * configure.ac:
- releasing 0.10.21, "We Could Be Wrong"
+2010-06-07 08:16:01 +0200 Martin Bisson <martin.bisson@gmail.com>
-2008-09-28 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for Y800 and Y16
+ Fixes bug #620441.
- * configure.ac:
- 0.10.20.4 pre-release
+2010-06-06 16:46:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix log function printf format issue
- Patch by: Vincent Penquerc'h <ogg dot k dot ogg dot k at googlemail dot com>
+2010-06-05 18:14:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
- Set the BOS flag on the BOS packet. Fixes #553244.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: stop jpeg typefinding once we found a SOF marker
-2008-09-23 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-05 18:05:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspmessage.c:
- (gst_rtsp_message_parse_request),
- (gst_rtsp_message_parse_response):
- Fix the g_return_val_if_fail() statements.
+ * tests/check/gst/typefindfunctions.c:
+ tests: fix memory leak in unit test
-2008-09-22 Michael Smith <msmith@songbirdnest.com>
+2010-05-19 15:40:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * 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.
-2008-09-15 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-06-05 17:22:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c:
- * gst/typefind/gsttypefindfunctions.c:
- Recognise Kate subtitle streams (#550582).
+ * 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
-2008-09-13 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-06-05 16:58:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
- Remove trailing comma from enum list, which causes problems
- with -pendantic (#550729).
+ * 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.
-2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-05 12:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- 0.10.20.2 pre-release
+ * gst/playback/gstscreenshot.c:
+ playbin2: improve screenshot code
+ Use appsrc and appsink in the screenshot code to make things nicer.
- * po/LINGUAS:
- * po/id.po:
- * po/pt_BR.po:
+2010-06-05 11:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- New translations.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix documentation string
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-06-05 11:05:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaysink.c:
+ playsink: add convert-frame action signal
+ Add a convert-frame action signal.
+ Fixes #620279
-2008-09-05 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 11:02:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/propertyprobe.c:
- Check for NULL pointer, in the hope that this fixes #532864.
+ * 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
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-06-05 10:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- No really, the next release is 0.10.21 (fix Since: tags in docs).
+ * 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
-2008-09-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-05 10:31:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaysink.c:
+ playsink: add property to get the last frame
+ Add a property to get the last video frame.
+ See #620279
-2008-09-04 Stefan Kost <ensonic@users.sf.net>
+2010-06-04 19:30:14 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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.
+
+2010-06-03 13:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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.
+
+2010-06-02 12:19:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: Allocate and free the clock in NULL->READY and reverse
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-01 23:49:07 -0700 David Schleef <ds@schleef.org>
- * 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
+ * common:
+ Automatic update of common submodule
+ From 17f89e5 to 47683c1
-2008-09-03 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+2010-06-01 22:54:33 -0700 David Schleef <ds@schleef.org>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
- Fixes #550638.
+ * common:
+ Automatic update of common submodule
+ From fd7ca04 to 17f89e5
-2008-09-03 Stefan Kost <ensonic@users.sf.net>
+2010-06-01 13:00:22 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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.
+ * 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.
+
+2010-05-31 19:28:45 +1000 Jonathan Matthew <jonathan@d14n.org>
+
+ * ext/gio/gstgiobasesink.c:
+ gio: map GIO NO_SPACE error to NO_SPACE_LEFT
+ Fixes bug #620140.
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+2010-05-28 08:27:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
- Disable subparse when xml is disabled. It woundn't work anyway. Fixes
- test runs.
+ * gst-libs/gst/interfaces/streamvolume.c:
+ configure: Remove (now) useless check for cbrt
-2008-09-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+2009-12-02 22:16:22 -0800 David Schleef <ds@schleef.org>
- * po/POTFILES.in:
- Add some more files with strings for translation.
+ * gst-libs/gst/interfaces/streamvolume.c:
+ interfaces: Use pow() instead of cbrt() for MSVC
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+2010-05-26 11:54:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c:
- Use new geo location tags from core. Fixes #481169
+ * common:
+ Automatic update of common submodule
+ From 357b0db to fd7ca04
-2008-09-01 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-05-26 08:51:09 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/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
-2008-08-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-05-24 11:27:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Jonathan Matthew <notverysmart gmail com>
+ * 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).
- * 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.
+2010-05-22 10:05:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate interlaced, chroma-site and color-matrix fields if necessary
- * 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.
+2010-05-22 10:02:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * gst/videorate/gstvideorate.c:
+ videorate: Use new string fixation function from core
- * 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.
+2010-05-22 09:48:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate color-matrix and chroma-site fields if necessary
- * 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.
+2010-05-22 09:39:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate the interlaced field if necessary
+ Fixes bug #619310.
- Patch by: Mike Ruprecht <cmaiku at gmail dot com>
+2010-05-22 08:55:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add IVF typefinder
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-21 18:16:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Alessandro Dessina <alessandro nnva org>
+ * gst/videorate/gstvideorate.c:
+ videorate: pass object to logging functions, use GST_DEBUG_FUNCPTR
- * 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.
+2010-05-20 15:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-08-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add basic descriptions for new WebM and VP8 types
- * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
- Reset variable so that query and convert fail after going back to
- READY. Fixes #548898.
+2010-05-20 14:21:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-22 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix sizes again, this time for real
- * 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.
+2010-05-20 13:58:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-21 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix size checks
- * 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.
+2010-05-20 10:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-21 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * tests/examples/seek/seek.c:
- Don't crash when having no visualisations.
+2010-05-20 08:52:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-16 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Some more minor adjustments for the VP8 Ogg mapping
- * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
- check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
- Fixes #548065.
+2010-05-19 21:35:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Update to the latest VP8 mapping
- * 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".
+2010-05-10 05:53:22 +0200 Philip Jägenstedt <philipj@opera.com>
-2008-08-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
- Add Lead H.264 here as well.
+2010-05-14 13:31:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-14 Julien Moutte <julien@fluendo.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix granulepos->key granule calculation for Dirac
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps): Add Lead H.264 variant.
+2010-05-14 11:02:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Add support for mapping specific granulepos to key granule mapping
- * 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.
+2010-05-05 13:59:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-12 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ ogg: Implement Ogg VP8 mapping
- * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
- since it has no basis in libtheora.
+2010-04-27 15:24:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-12 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for On2 VP8
- * gst-libs/gst/interfaces/propertyprobe.h:
- Remove double "interface" from doc-string.
+2010-05-19 16:17:19 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
- * gst-libs/gst/interfaces/xoverlay.h:
- Document interface.
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix a typo introduced by 9d753824.
+ video/x-raw-float => audio/x-raw-float. Fixes #619090.
- * gst-libs/gst/riff/riff.c:
- Add basic doc blobs.
+2010-05-18 08:45:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * gst-libs/gst/audio/Makefile.am:
- Don't try to build that example anymore.
+2010-05-15 21:15:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ configure: Use = instead of == in shell scripts for equality checks
- * 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.
+2010-05-14 18:23:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ Automatic update of common submodule
+ From 4d67bd6 to 357b0db
- * 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.
+2010-05-14 17:24:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Always take the object lock when accessing the caps
+ Fixes bug #618625.
- * 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.
+2010-05-14 17:17:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-11 Stefan Kost <ensonic@users.sf.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.
- * gst-libs/gst/audio/audio.c:
- Fix doc comment syntax.
+2010-05-14 17:13:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/interfaces/propertyprobe.c:
- Add more doc-comments and a FIXME: for the signal.
+ * gst/playback/gstplaysink.c:
+ playsink: Fix deadlock caused from an additional lock instead of unlock
+ Also improve debug output for the playsink lock.
-2008-08-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-13 12:16:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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>
+ * 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.
- * gst-libs/gst/pbutils/descriptions.c:
- Add audio/x-qdm for qtdemux.
+2010-05-13 12:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-08-05 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * ext/vorbis/vorbisdec.c:
- Do not leak old taglist.
+2010-05-13 08:05:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-04 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/video/video.c:
+ video: Use simple fraction multiplication functions instead of going through GValues
- * tests/icles/test-scale.c:
- Include <stdlib.h> for atoi().
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-04 Andy Wingo <wingo@pobox.com>
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a unit test for checking if the negotiation works as expected
- * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
- documentation fix.
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try harder to keep the DAR if possible
+ Fixes bug #371108.
- * 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.
+2010-05-10 15:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Log PAR and DAR of input and output caps when setting caps
+
+2010-05-10 14:52:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Set input width/height if the output caps don't have any width or height
+
+2010-05-10 13:01:44 +0200 Andoni Morales <ylatuya@gmail.com>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try to keep DAR when scaling
+ Fixes bug #371108.
+
+2010-05-10 19:09:28 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * 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
+
+2010-05-10 18:53:29 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ basertpaudiopayload: Set duration on buffers
+ Set the duration of the buffers from their size
+
+2010-05-11 16:12:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate PAR to 1/1 if possible
+
+2010-05-11 10:07:01 +0200 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.
+ configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0
-2008-08-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-10 12:44:41 +0200 Sebastian Dröge <sebastian.droege@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'.
+ * 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.
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-10 12:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
- Add some more comments.
+ * sys/ximage/ximagesink.c:
+ ximagesink: Post an error message on the bus if no supported pixmap formats can be found
+ Might fix bug #615851.
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-07 19:49:57 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace : whooops
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-07 19:21:13 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-scale.c: (make_pipeline), (main):
- Add dynamic rescaling tests for the new basetransform.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: more minor cleanups
-2008-07-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-05-07 17:16:28 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/audioconvert/Makefile.am:
- Dist recently-added gstfastrandom.h.
+ * 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)
-2008-07-30 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-05-07 12:19:25 +0200 Edward Hervey <bilboed@bilboed.com>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Fix a "may be used uninitialized in this function" which weirdly only
- appears on macosx (?).
+ * 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
-2008-07-30 Stefan Kost <ensonic@users.sf.net>
+2010-05-06 18:50:51 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/riff/riff-ids.h:
- Adding acid chunk for tempo and loop information.
+ * 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>
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: rename are_raw_caps to are_final_caps, correct comment
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
- * sys/xvimage/Makefile.am:
- floor() needs linking to $(LIBM).
+2010-05-07 17:16:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ sdpmessage: add new TIAS bandwidth modifier
+ Add TIAS modifier as specified in RFC 3890.
+ Do some whitespace fixes.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Aggregate short reads and add some comments and debug logging.
- Fixes #537380
+2010-05-07 00:10:22 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: disambigue comment due to popular demand
+ Write "target depth" instead of "our depth" or previous ambigous "out depth".
- * gst/playback/gstplaybasebin.c:
- Fix property doc markup (its not a signal).
+2010-05-06 15:40:34 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Add since tag for new proeprties (also add sice tags fro the last two
- other additions).
+ * gst/playback/gstplaysink.c:
+ playsink: disconnect signals in some more cleanup cases
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2010-05-06 13:10:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add autofill/colorkey properties. Fixes #538656.
+ * 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.
+
+2010-05-06 12:21:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2010-05-06 12:06:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2010-05-06 10:56:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2010-05-06 11:34:53 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ docs: be more firendly to gtk-doc limitted parsing capabilities
-2008-07-28 David Schleef <ds@schleef.org>
+2010-05-06 09:42:02 +0300 Stefan Kost <ensonic@users.sf.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
+
+2010-05-06 09:17:33 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: use can_intersect to avoid a caps copy
+
+2010-05-06 09:14:25 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * 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 :/.
+
+2010-05-06 09:12:32 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstdecodebin.c:
+ decodebin: use can_intersect to avoid a caps copy
+
+2010-05-06 09:11:17 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * 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.
+
+2010-05-06 08:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * 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.
+
+2010-05-06 08:16:45 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: fix typo in comment
+
+2010-05-06 08:15:16 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/ximage/ximagesink.c:
* 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.
+ 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.
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-15 13:09:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-15 13:08:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggmux.c:
- Allow muxing of CELT into Ogg streams.
+ * tests/icles/audio-trickplay.c:
+ example: update status (adder is fixed now)
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-15 13:08:01 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
- (plugin_init):
- Add simple typefinder for the CELT codec (www.celt-codec.org).
+ * tests/icles/playbin-text.c:
+ example: make app static
-2008-07-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-05 13:25:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Jan Gerber <j at oil21 dot org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: printf format fixes
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
- Fix calculation of the start time from skeleton streams.
- Fixes bug #530068.
+2010-05-04 15:32:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-07-24 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * tests/examples/seek/seek.c:
- Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
+2010-05-04 17:54:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * 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.
+2010-05-04 17:38:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-23 Michael Smith <msmith@songbirdnest.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: handle corrupt indexes
+ Make sure we handle and receover from corrupt indexes.
- * configure.ac:
- Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
+2010-05-04 15:47:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-23 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix EOS check
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+2010-05-04 13:51:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoild division by 0
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-05-04 13:50:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: cleanup unused defines
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-05-04 13:36:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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):
+ * 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.
- 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.
+2010-05-04 13:05:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-16 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use skeleton duration when possible
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Add 'ticks', a 1/30 second sine wave pulse every second.
+2010-05-04 13:02:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-15 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggstream: parse duration from 3.3 skeleton
- * gst-libs/gst/video/video.c: Revert ABI change.
+2010-03-02 11:16:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: more index parsing work
- * 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.
+2010-03-01 13:50:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-14 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * 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).
- * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
- Video format can also be conveniently determined from (many)
- non-fixed caps.
+2010-05-04 11:19:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-14 Jan Schmidt <thaytan@noraisin.net>
+ * 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.
- * 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.
+2010-05-03 16:52:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-11 Stefan Kost <ensonic@users.sf.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/audioconvert/gstaudioconvert.c:
- Remove now obsolete note in the docs.
+2010-04-30 18:37:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-11 Stefan Kost <ensonic@users.sf.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.
- * 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.
+2010-04-30 18:03:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: refactor for seeking in pushmode
+ refactor the code a little to prepare for seeking in push mode
- * tests/examples/seek/Makefile.am:
- Fix out of tree build by adding all required CFLAGS.
+2010-05-03 12:46:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * 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.
+2010-04-26 22:08:41 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * 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.
+2010-04-30 19:56:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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
- Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
+2010-04-30 19:37:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: ensure writable metadata
- decodebin2 doesn't have this issue because all streams of a group
- go through multiqueue.
+2010-04-30 17:41:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-07-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+2010-04-30 14:07:59 +0100 Tim-Philipp Müller <tim.muller@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.
+ * configure.ac:
+ * ext/gio/gstgiobasesrc.c:
+ Bump GLib requirement to 2.20
+ See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
-2008-07-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-30 13:36:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@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/playback/gstplaybin2.c:
+ playbin2: forward duration query duration during group switch if no cached duration
+ ... such as during first group setup.
+ Fixes #616396.
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-02 16:37:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: only EOS when all streams are EOS
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-02 16:36:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix debug message
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-04-30 08:45:43 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * tests/icles/audio-trickplay.c:
+ test: fix copy and paste error of variable name
-2008-06-29 Stefan Kost <ensonic@users.sf.net>
+2010-04-18 20:46:37 +0300 Stefan Kost <ensonic@users.sf.net>
* gst/adder/gstadder.c:
- Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
-
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
-
- * ChangeLog:
- ChangeLog surgery.
-
- * tests/examples/seek/seek.c:
- Move variable into ifdef too.
-
-
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
-
- * tests/examples/seek/seek.c:
- Include config.h and check if we have X. Remove XInitThread(), don't
- think its needed. Fixes: #540334.
-
-2008-06-26 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- 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 Stefan Kost <ensonic@users.sf.net>
-
- * tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
- Remove libgstvideo usage. Use gtk_get_option_group instead of
- gtk_init().
-
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/Makefile.am:
- Name the test registry format neutral.
-
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
-
- * gst/playback/gstqueue2.c:
- Do not double notify. Remove the unsued return value.
+ adder: only accept seek-types none and set
+ Previously we were also acting on cur and end, but treating them like none.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2010-04-14 23:31:20 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2010-04-26 17:30:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
- Embedd the xwindow. Allow to play multiple files.
+ * 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.
-2008-06-24 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-26 11:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ playbin2: Add flag for enabling/disabling automatic deinterlacing
-2008-06-22 Stefan Kost <ensonic@users.sf.net>
+2010-04-26 11:11:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c:
- Fix crasher when playing a parse-launch line the 2nd time.
+ * gst/playback/gstplay-enum.c:
+ playbin: Use g_once_init_{enter,leave} instead of GOnce for enum/flag registration
-2008-06-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-04-23 17:01:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/oggmux.c:
- Properly ifdef tests to fix compilation.
+ * 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.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-04-23 16:24:11 +0200 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.
+ * gst/playback/gstplaysink.c:
+ playsink: Correctly reconfigure the video chain when switching from a subtitle to a non-subtitle file
+ Fixes bug #616422.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-04-23 16:08:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaybin2.c:
- Use a different constant for the convert-frame signal id.
- Fixes #537009.
+ 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.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-04-23 15:30:54 +0200 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.
+ playbin2: Set subtitle encoding on the decodebins again
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-04-23 15:22:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-04-23 15:14:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin2.c:
- Ensure decodebin2 emits 'drained' signal once, and only once, when all
- pads are drained.
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Some random cleanup
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-23 15:06:13 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for Y444, Y42B and Y41B
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-23 14:42:50 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Reorder template caps by the amount of information contained in the color formats
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-22 15:46:17 -0400 Joshua M. Doe <joshua.doe@us.army.mil>
- * 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.
+ * gst/videorate/gstvideorate.c:
+ videorate: add support for video/x-raw-gray
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-29 15:05:59 +0100 Tim-Philipp Müller <tim.muller@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.
+ * 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).
-2008-06-20 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-29 15:02:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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>
+ * 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.
- * configure.ac:
- releasing 0.10.20, "Here I Go Again"
+=== release 0.10.29 ===
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-28 02:16:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- 0.10.19.3 pre-release
+ * 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
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-28 01:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Fix build on win32.
- Patch By: David Schleef <ds@schleef.org>
- Fixes: #536874
+ * 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
-2008-06-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-25 23:14:35 +0100 Tim-Philipp Müller <tim.muller@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.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.28.3 pre-release
+
+2010-04-20 17:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-plugins-base.doap:
+ doap: update repository info from cvs->git and maintainers
- 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.
+2010-04-23 14:39:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Fixes bug #536849 and #536848.
+ * common:
+ Automatic update of common submodule
+ From fc85867 to 4d67bd6
- * 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.
+2010-04-22 20:58:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-05 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y41B->Y444 conversion
+ ...which is the intermediate conversion for conversion to all
+ other formats.
+ Fixes bug #616545.
+
+2010-04-16 20:03:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Don't leak the input buffer in error cases
+ Fixes bug #615572.
+
+2010-03-29 12:53:11 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/ogg/gstoggmux.c:
+ docs: fix typo in link name
+
+2010-04-15 12:59:53 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * 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
+
+2010-04-15 11:44:49 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: refactor _update_geometry()
+ Refactor like in xvimagesink. Remove the extra parameter and adjust the assert check.
+
+2010-04-15 07:18:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
* configure.ac:
- 0.10.19.2 pre-release
+ 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
-2008-06-04 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-14 14:13:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstrtsp.def:
- * win32/common/libgsttag.def:
- Add new API functions to the dll exports
+ * 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
-2008-06-04 Michael Smith <msmith@songbirdnest.com>
+2010-04-14 13:50:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-13 16:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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).
+ * 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.
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-04-14 13:43:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Do not try to run the check tests for subparse unless it has been
- built.
+ * 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.
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-04-12 14:02:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add channels and rate to ADTS caps if we can
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-04-12 13:33:18 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * 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.
+ * 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>
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+ * 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/tcp/gstmultifdsink.c: (setup_dscp_client):
- Fixed accidental use of IPv4 options for all IPv6 addresses.
+ * 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.
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-26 19:00:47 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/interfaces/mixertrack.h:
- Document mixer track flags.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: use ptime-multiple
+ Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>
+ Fixes #613248
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-09 16:06:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * 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
- * 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.
+2010-04-09 11:20:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 218568f to fc85867
- * 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.
+2010-04-08 17:49:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.h:
+ playback, ogg: dist new gstplayback.h and gstogg.h
- * 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.
+2010-04-09 08:23:33 +0200 Thomas Green <thomasgr33n@googlemail.com>
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- Based on patch by: John Millikin <jmillikin gmail com>
+2010-04-08 07:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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-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.
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-08 12:21:50 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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...).
+ * ext/gio/gstgio.c:
+ * ext/gnomevfs/gstgnomevfs.c:
+ ext: Invert rank of gio and gnomevfs elements
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-08 01:26:09 +0100 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
+ * 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.
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-07 20:21:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add gst_audio_check_channel_positions() to the exported symbols.
+ * gst-libs/gst/video/video.h:
+ docs: add gtk-doc chunks with Since: tags for new GST_VIDEO_CAPS_GRAY* API
+
+2010-04-07 19:07:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * 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.
+
+2010-04-07 17:25:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/video/video.c:
+ video: Fix parsing of 8-bit grayscale caps
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-07 17:21:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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-libs/gst/video/video.h:
+ video: API: Add GST_VIDEO_CAPS_GRAY{8,16}
- * tests/check/libs/audio.c: (GST_START_TEST):
- Add some simple checks for gst_audio_check_channel_positions().
+2010-04-07 17:08:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ video: API: Add gst_video_format_is_gray() to the docs
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- minrange and maxrange are scaled according to the frequency
- multiplier.
+2010-04-07 17:07:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * win32/common/libgstvideo.def:
+ video: Add new symbol to the exported symbols list
- * 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).
+2010-04-07 17:06:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
- (main):
- Test with odd height/width too.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for 8-bit and 16-bit grayscale formats
-2008-06-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-06 10:55:42 +0200 Wim Taymans <wim.taymans@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.
+ * 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
-2008-05-31 Mark Nauwelaerts <mnauw@users.sf.net>
+2010-04-04 21:24:44 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * configure.ac:
+ configure: fix cdparanoia check
+ Linking with libcdda_paranoia.so requires also linking with
+ libcdda_interface.so.
-2008-05-31 Mark Nauwelaerts <mnauw@users.sf.net>
+2010-04-04 18:00:23 -0300 Thiago Santos <thiago.sousa.santos@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>
+ * 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/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>
+2010-04-04 21:18:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * win32/common/libgstaudio.def:
- Add gst_audio_clock_reset to the list of exported symbols.
+ * 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.
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-04 20:55:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
- Remove wrong_channels_identification_header unit test as we now
- support 7 (and more channels).
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: Detect errors from device polling, stop the task and post an error message
+ Partially fixes bug #614545.
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-04 12:13:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-03 22:49:11 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
- Allow setting channel positions for 1/2 channels when using
- gst_audio_set_structure_channel_position().
+2010-04-03 14:03:45 +0100 Vincent Untz <vuntz@gnome.org>
- * 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.
+ * 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.
- We now allow previously conflicting channel positions to appear
- together (rear center and rear left/right for example).
- Fixes bug #533817.
+2010-04-02 21:01:25 +0300 Stefan Kost <ensonic@users.sf.net>
- 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.
+ * 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/audioconvert.c: (GST_START_TEST):
- Adjust unit tests with non-standard 1/2 channel layouts to the more
- correct new behaviour.
+2010-04-02 20:56:19 +0300 Stefan Kost <ensonic@users.sf.net>
- Add a unit test for 5.1->Stereo downmixing.
+ * 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.
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-02 18:56:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/examples/seek/seek.c:
+ seek: Only use embed_xid if HAVE_X is defined
+ Fixes bug #614622.
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-01 19:13:22 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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.
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-01 18:51:17 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ tests: Unref the bus once we're done with it
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-01 16:49:37 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * common:
+ common: Update for new suppressions
- * 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.
+2010-04-01 13:55:15 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-28 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/playback/gstplaysink.c:
+ gstplaysink: Remove unused variable.
+ The value of klass is never used
- * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
- Initialise error to NULL as we should.
+2010-04-01 13:53:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-28 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Removing dead assignment.
+ The value of group is overwritten a few lines below before being used.
- * gst/adder/gstadder.c: (gst_adder_query_duration),
- (gst_adder_query_latency), (gst_adder_query):
- Implement latency query.
+2010-04-01 13:51:13 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-27 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Remove unused variable
- * gst/adder/gstadder.c: (gst_adder_query_duration):
- Correctly resync the iterator if gst_iterator_next() returns
- GST_ITERATOR_RESYNC.
+2010-04-01 13:48:42 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/gnomevfs/gstgnomevfssink.c:
+ gstgnomevfssink: Return the proper GstFlowReturn.
+ We were always returning GST_FLOW_OK previously even if we encountered errors.
- * win32/vs6/libgstpbutils.dsp:
- Add pbutils-enumtypes.c to sources (#518037).
+2010-03-30 23:44:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-27 Wim Taymans <wim.taymans@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:
+ 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.
- * 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()
+2010-03-30 20:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * tests/examples/seek/jsseek.c:
+ examples: fix printf format warning in jsseek example
+ Yes, I know about G_GSIZE_FORMAT.
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-30 19:56:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-30 19:29:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/pipelines/oggmux.c: (test_pipeline):
- Don't use deprecated function.
+ * gst-libs/gst/Makefile.am:
+ Our RIFF library depends on both the audio and tag libraries
+ Update rules in Makefile.am accordingly.
-2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-30 15:10:42 +0200 Robert Swain <robert.swain@collabora.co.uk>
- * 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/gstplaysink.c:
+ playsink: Fix aduio_raw_sink typo
- * gst/playback/gstplaybin2.c: (perform_eos):
- When pushing out the EOS event, collect the return value and warn when
- something failed.
+2009-11-28 21:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
-2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/examples/seek/.gitignore:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/jsseek.c:
+ examples: Add a silly joystick based shuttle example
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for DVCPRO.
+2010-03-29 20:07:52 -0700 David Schleef <ds@schleef.org>
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: 0-length packets are delta units
- * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
- Change default scaling method from nearest-neighbour to bilinear.
+2010-03-29 10:47:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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/check/libs/video.c:
- More checks.
+2010-03-29 00:22:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-25 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- * 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.
+2010-03-28 21:50:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-23 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/Makefile.am:
+ gst-libs: specify dependencies in Makefile.am to make them explicit
- * 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.
+2010-03-24 09:59:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-22 Jan Schmidt <jan.schmidt@sun.com>
+ * 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>
- * 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:
+ uridecodebin: we can handle avi in download mode too
+ Add avi to the whitelisted types that can be used for download buffering.
- 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.
+2010-03-26 15:57:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- 100% symbol coverage for the plugin docs, booya.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Provide packet duration function for old FLAC mapping too
+ Fixes bug #613809.
-2008-05-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-18 22:12:40 +0000 Damien Lespiau <damien.lespiau@intel.com>
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * 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/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
- Fix generation of NV12/NV21 frames. Fixes bug #532454.
+2010-03-24 23:04:43 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+2010-03-24 14:43:21 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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.
+ * 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.
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
+2010-03-24 10:18:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * docs/Makefile.am:
- Fix installing plugin documentation when gtk-doc is disabled.
+ * 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
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
+2010-03-23 09:32:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtsp/Makefile.am:
- Distribute, don't install md5.h
+ * 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>
-2008-05-21 Julien Moutte <julien@fluendo.com>
+ * 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.
- * 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.
+2010-03-24 12:10:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ rtsptransport: ignore unparsable ranges
+ Ignore unparsable port ranges instead of erroring out.
+ Fixes #613591
- * 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>
+2010-03-23 18:36:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/libgstrtsp.def:
+ win32: Add new gst_rtsp_lower_trans_get_type() symbol to the symbol lists
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-23 11:01:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * docs/design/part-playbin2.txt:
- Add some leftover doc.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add some more fourcc for MPEG-4 video
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-22 09:15:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
- Fix copy & paste error in last commit.
+ * configure.ac:
+ configure: require core git
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-22 08:38:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft.pc.in:
+ pkgconfig: Add @LIBM@ to the FFT pkg-config files
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-22 08:35:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Henrik Eriksson <henriken at axis dot com>
+ * 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-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
- * 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.
+2010-03-20 13:42:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- * 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.
+2010-03-20 12:55:36 +0000 Torsten Schönfeld <kaffeetisch@gmx.de>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/interfaces/xoverlay.c:
+ docs: add Since: tags to gst_x_overlay_handle_event() docs
+ Fixes #613403.
- * 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.
+2010-03-19 22:33:58 +0100 Benjamin Otte <otte@redhat.com>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Constify some strings in the API
+ Needed by plugins-good
- * gst/audioconvert/gstchannelmix.c:
- (gst_channel_mix_fill_normalize):
- Prevent division by zero if the channel mix matrix contains only
- zeroes.
+2010-03-19 16:41:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+2010-03-19 15:37:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
- Close a buffer memory leak. Fixes bug #534071.
+ * 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.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-19 15:00:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
* 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.
+ rtsp: add GType for transport flags
+ Make a method to register the transport flags as a GType.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-19 01:00:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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).
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-19 00:46:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore new unit test binary
-2008-05-20 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-17 23:57:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- Require core CVS for GstBaseSrc buffer caps setting magic.
+ configure.ac: -Wmissing-prototypes and -Wnested-externs are not valid for C++
+ Fixes building Qt-based overlay examples in combination with -Werror.
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-17 16:32:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Fix logic in last commit.
+ * configure.ac:
+ configure.ac: wrap overly long warning flag lines
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-17 19:24:27 -0300 Reuben Dowle <reube.dowle@navico.com>
- * 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.
+ * sys/ximage/ximagesink.c:
+ ximagesink: Fix caps leak
+ Unref caps when peer doesn't accept caps
+ Fixes #613198
- * 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.
+2010-03-17 08:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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>
+2010-03-17 16:46:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Correctly set the default channel positions when converting to 8
- channels.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Use correct boundary checks for YUY2/UYVY
+ Fixes bug #613093.
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-17 16:39:13 +0100 Peter Kjellerstedt <peter.kjellerstedt@axis.com>
- * configure.ac:
- Error out if we don't have the required version of core.
+ * 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.
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-17 15:41:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/volume/gstvolume.c:
+ volume: Remove useless cast
+ It's not necessary anymore after latest core change to GstValueArray.
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-17 12:08:30 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * 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.
-2008-05-16 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-17 11:14:29 +0100 Benjamin Otte <otte@redhat.com>
- * gst/adder/gstadder.c:
- Adder doesn't support 24 bit samples so don't claim it supports them
- in the pad template caps.
+ * 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.
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-17 10:53:21 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * configure.ac:
+ * 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.
- * 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.
+2010-03-17 10:51:57 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * 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
-2008-05-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-17 10:47:07 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * 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.
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-16 20:05:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Bernard B <b-gnome at largestprime dot net>
+ * 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
- * 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.
+2010-03-16 18:31:15 +0100 Benjamin Otte <otte@redhat.com>
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add a testcase for seqnum compare function.
+ * ext/alsa/gstalsaplugin.c:
+ * ext/ogg/gstoggdemux.c:
+ More ENABLE_NLS fixes
-2008-05-14 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-16 18:06:16 +0100 Benjamin Otte <otte@redhat.com>
- * 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().
+ * gst-libs/gst/gettext.h:
+ Fix for ENABLE_NLS being undefined for -Wundef
-2008-05-14 Stefan Kost <ensonic@users.sf.net>
+2010-03-15 22:49:53 +0100 Benjamin Otte <otte@redhat.com>
+ * 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:
- Better debug logging in port value handling. Merging separate port
- value loops into one.
-
-2008-05-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- Patch by: Hannes Bistry <hannesb at gmx dot de>
+2010-03-16 15:45:23 +0100 Benjamin Otte <otte@redhat.com>
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
+ * 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_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 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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/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.
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-15 22:17:56 +0100 Benjamin Otte <otte@redhat.com>
- * win32/common/libgstrtsp.def:
- Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
- symbols.
+ * configure.ac:
+ Add -Wundef flag
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-16 16:15:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: allow for more ipv6 addresses
+ Use hints in getaddrinfo() so that we can also resolve ipv6 addresses.
- * 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.
+2010-03-11 14:52:09 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
+ Fixes #612223.
- * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
- Fix nv12<->nv21 conversion if stride is larger than width.
+2010-03-16 01:08:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Jan Gerber <j at oil21 dot org>
+2010-03-15 12:54:32 -0500 Rob Clark <rob@ti.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP7 fourccs
+ Fixes #612968.
-2008-05-12 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 12:54:01 -0500 Rob Clark <rob@ti.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP62 fourcc
+ See #612968.
-2008-05-11 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * gst-libs/gst/pbutils/install-plugins.c:
- Fix docs: type and missing word.
-
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * 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.
-
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * 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.
-
-2008-05-09 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-15 23:46:39 +0000 Tim-Philipp Müller <tim.muller@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/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/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 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
- Choose to allocate one less segment but require one additional segment
- as latency.
+2010-03-15 23:47:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
- No need to increment the number of segments in the source.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspextension.c:
+ docs: add GstRTSPExtension to docs
+ Add minimal docs for GstRTSPExtension so people know it exists.
- * 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>
+2010-03-15 18:45:13 +0000 David Hoyt <dhoyt@llnl.gov>
- * 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.
+ * 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.
-2008-05-08 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-03-15 13:39:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst-libs/gst/tag/gstxmptag.c:
+ docs: fix typo in gst_tag_list_from_xmp_buffer() docs chunk
-2008-05-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 13:32:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Wouter Cloetens <zombie at e2big dot org>
+ * 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
- * 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.
+2010-03-15 13:24:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: add new libgstvideo API to documentation
-2008-05-08 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-15 13:19:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * 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()
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Let audioresample use the buffer allocation of basetransform instead
- of it's own stuff.
+2010-03-15 13:14:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: use C comments instead of C++-style comments
-2008-05-07 Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
+2010-03-15 13:10:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: use g_value_set_static_string() for string constants
-2008-05-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 14:26:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_query):
- Report the latency with the new seglatency parameter.
+ * gst/playback/gstplaysink.c:
+ playsink: Avoid g_object_set() on NULL if a text sink is used
+ Fixes bug #611702.
- * 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.
+2010-03-15 14:10:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/subparse/gstsubparse.c:
+ subparse: Correctly escape brackets in DKS regex
+ Fixes bug #612783.
- * 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.
+2010-03-15 11:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-15 11:41:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
- * 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.
+2010-03-15 01:35:15 -0700 David Schleef <ds@schleef.org>
-2008-05-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * gst/playback/gstdecodebin2.c: (connect_pad):
- When autoplugging fails, set the element back to NULL before
- unreffing it.
+2010-03-15 01:34:09 -0700 David Schleef <ds@schleef.org>
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/theora/gsttheoradec.c:
+ theoradec: add chroma-site to caps
- * win32/common/libgstaudio.def:
- Add gst_base_audio_src_[sg]et_slave_method() to the exported
- symbols.
+2010-03-15 01:33:36 -0700 David Schleef <ds@schleef.org>
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: add chroma-site to caps
- * 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.
+2010-03-15 01:31:20 -0700 David Schleef <ds@schleef.org>
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add a simple unit test for SAMI parsing.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: add gst_video_parse_caps_chroma_site()
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-14 19:10:16 -0700 David Schleef <ds@schleef.org>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * ext/theora/gsttheoradec.c:
+ theoradec: add color-matrix to caps
- * 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.
+2010-03-14 16:17:46 -0700 David Schleef <ds@schleef.org>
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add color-matrix to template caps
- * 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.
+2010-03-14 22:14:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-05 Edward Hervey <edward.hervey at collabora co uk>
+ * 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.
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- It's SorensOn and not SorensEn.
+2010-03-14 22:13:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * Makefile.am:
+ build: add cruft alert for common/shave* leftovers to top-level Makefile.am
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Fix description of video/x-flash-video.
+2010-03-14 13:11:53 -0700 David Schleef <ds@schleef.org>
-2008-05-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't drop zero-sized packets
+ Zero-sized packets have relevence to Theora.
- * 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.
+2010-03-12 15:47:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_free_noise_shaping):
- Don't return before freeing the noise shaping history.
+ * 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.
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-12 15:44:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/subparse.c: (do_test),
- (test_tmplayer_style3b), (subparse_suite):
- Add unit test for the tmplayer variant from bug #530962.
+ * gst/volume/gstvolume.c:
+ volume: Only allocate a mute value array if a control source exists for the mute property
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-12 13:55:55 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From e272f71 to 55cd514
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-10 10:50:32 -0300 Thiago Santos <thiago.sousa.santos@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-libs/gst/tag/gstxmptag.c:
+ tags: Add new mapping to XMP helpers
+ Adds geotagging mappings to XMP helpers
+ Fixes #609539
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-11 20:16:44 +0100 Benjamin Otte <otte@redhat.com>
- * 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-libs/gst/interfaces/Makefile.am:
+ Don't have 2 include dirs
+ Seems to have been accidentally introduced in
+ 7269bc26d0a4bf44bd77a039fb54777625ef5f39.
-2008-05-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 16:35:10 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * tests/icles/audio-trickplay.c:
+ tests: Fix another unitialized variable
-2008-05-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 16:09:26 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * tests/icles/audio-trickplay.c:
+ tests: Fix unitialized variable.
-2008-05-02 Stefan Kost <ensonic@users.sf.net>
+2010-03-11 15:38:18 +0100 Benjamin Otte <otte@redhat.com>
- * 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 :)
+ * 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.
-2008-05-02 Thijs Vermeir <thijsvermeir@gmail.com>
+2010-03-11 13:32:14 +0100 Benjamin Otte <otte@redhat.com>
- * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
- (paint_setup_NV21), (paint_hline_NV12_NV21):
- Add support for NV12 and NV21 in videotestsrc
+ * 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.
-2008-05-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-11 12:49:02 +0100 Benjamin Otte <otte@redhat.com>
- * 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/gio/gstgiobasesink.c:
+ gio: Remove unused function
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-11 11:14:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/vorbis/gstvorbisparse.c:
+ vorbisparse: make sure header buffer metadata is writable before modifying it
+ Fixes unit test failures with core git.
-2008-04-30 Michael Smith <msmith@songbirdnest.com>
+2010-03-11 12:18:00 +0100 Benjamin Otte <otte@redhat.com>
- * ext/theora/theoradec.c:
- Cool kids don't divide by zero.
- Treat PAR of x:0 as 1:1.
- Fixes #530719.
+ * 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.
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-11 12:04:32 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * configure.ac:
+ Add WARNING_CXXFLAGS where ERROR_CXXFLAGS are
+ This matches the previous commit doing the same for CFLAGS in response
+ to the common/ module changes.
-2008-04-28 Michael Smith <msmith@songbirdnest.com>
+2010-03-11 12:04:37 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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
-2008-04-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 10:38:53 +0000 Tim-Philipp Müller <tim.muller@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()
+ * 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.
-2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 11:20:48 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Fix the docs about the seqnum compare function, it returns a difference.
+ * common:
+ Automatic update of common submodule
+ From df8a7c8 to e272f71
-2008-04-24 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-03-11 10:55:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsadeviceprobe.c:
- (gst_alsa_get_device_list): Don't return before freeing up
- the allocated structures.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xvoverlay: correct version number in docs
-2008-04-24 Stefan Kost <ensonic@users.sf.net>
+2010-02-26 13:56:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c:
- Remove obsolete streaminfo code and fix a leak. Fixes #529546
+ * 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.
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-02-08 17:20:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggdemux.c:
- Revert the event part, that should not go in.
+ * 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:
+ 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.
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-02-19 14:38:36 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/ogg/gstoggdemux.c:
- Don't leak GstPluginFeatures when filtering.
+ * 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.
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+2010-02-19 14:46:43 +0200 Stefan Kost <ensonic@users.sf.net>
* sys/xvimage/xvimagesink.c:
- Add some logging for cases when grabbing the xv failed.
+ * 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.
-2008-04-21 David Schleef <ds@schleef.org>
+2010-02-17 15:00:13 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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()
-2008-04-21 David Schleef <ds@schleef.org>
+2010-02-16 12:06:08 +0200 Stefan Kost <ensonic@users.sf.net>
+ * sys/ximage/ximagesink.c:
* 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.
+ 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.
-2008-04-21 Stefan Kost <ensonic@users.sf.net>
+2010-03-10 21:51:59 +0100 Benjamin Otte <otte@redhat.com>
- * ext/gnomevfs/gstgnomevfsuri.c:
- Add FIXME comment about using uri-list for source and sink.
+ * common:
+ Automatic update of common submodule
+ From 9720a7d to df8a7c8
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 21:01:20 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * 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.
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 20:43:46 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 0b6e072 to 9720a7d
- Add locking to the properties and use
- gst_element_class_set_details_simple() instead of a static
- GstElementDetails struct.
+2010-03-10 16:09:45 +0100 Benjamin Otte <otte@redhat.com>
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 7cc5eb4 to 0b6e072
- * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
- (plugin_init):
- Add "mpp" and "mp+" as possible extensions for MusePack files.
+2010-03-10 14:36:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Add typefinding for MusePack StreamVersion 8 files and include the
- stream version in the caps.
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: do not cache FLUSH_START/_STOP events
+ ... and similarly so for serialized events.
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 14:34:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@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/playback/gstplaysink.c:
+ playsink: provide correct error message if configured audio/video sink fails
-2008-04-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 10:22:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * ext/vorbis/gstvorbisdec.h:
+ vorbisdec: remove unused field
-2008-04-18 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-02-02 11:34:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Don't validate the payload if there isn't any.
- Fixes #525915
+ * tests/check/pipelines/vorbisdec.c:
+ tests: enable strict discontinuity checking on vorbisdec pipeline
+ Closes #423086.
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 01:09:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
- Use g_atomic_int_set() instead of gst_atomic_int_set().
+ * common:
+ Automatic update of common submodule
+ From 7aa65b5 to 7cc5eb4
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 01:07:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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-libs/gst/video/video.c:
+ docs: fix Returns: for gst_video_parse_caps_color_matrix()
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-03-10 00:46:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Remove cpp style commented old code.
+ * 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
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-03-10 00:42:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin2.c:
- Fix signal docs.
+ * 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.
-2008-04-14 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 00:41:13 +0000 Tim-Philipp Müller <tim.muller@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.
+ * tests/check/elements/videorate.c:
+ tests: don't use deprecated functions in videorate unit test
-2008-04-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 00:29:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * win32/common/libgstvideo.def:
+ win32: add new API to libgstvideo.def
- * tests/check/libs/video.c:
- Fix unit test not to create caps with width=0 and height=0.
+2010-03-09 15:39:55 -0800 David Schleef <ds@schleef.org>
-2008-04-11 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggmux.c:
+ oggmux: Don't flush after every frame for theora
- * docs/design/draft-keyframe-force.txt:
- Fix typo.
+2010-03-09 21:26:58 +0000 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 44ecce7 to 7aa65b5
+
+2010-03-09 13:05:23 -0800 David Schleef <ds@schleef.org>
+
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add color-matrix handling to caps
+
+2010-01-30 22:55:01 -0800 David Schleef <ds@schleef.org>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Add color-matrix to caps
+
+2010-02-26 16:25:59 +0100 Sebastian Dröge <sebastian.droege@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:
+ * pkgconfig/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tools/Makefile.am:
+ build: Make some more rules silent if requested
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_handle_src_query):
- Set buffering mode in the messages.
- Set buffering percent in the query.
+2010-02-26 15:40:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-23 19:12:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-17 19:18:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ volume: If a controller is used, use sample accurate property values
+ Fixes bug #609801.
- * 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.
+2010-03-09 19:17:04 +0100 Benjamin Otte <otte@redhat.com>
-2008-04-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/video/video.c:
+ gstvideo: Fix typos in comments
- * 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.
+2010-03-09 17:32:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-07 Jan Schmidt <jan.schmidt@sun.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-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
- * 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.
+=== release 0.10.28 ===
-2008-04-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-08 23:20:43 +0000 Tim-Philipp Müller <tim.muller@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-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:
- Merge other changes from 0.10.19 release branch.
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.28
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-08 23:19:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-08 21:57:03 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- Remove some more fields.
+ * 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
-2008-04-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-08 14:50:25 +0000 David Schleef <ds@schleef.org>
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * ext/ogg/dirac_parse.c:
+ oggdemux: fix dirac header parsing
+ Fixes #611900.
- * configure.ac:
- Actually build dlls when cross-compiling with mingw32.
- Fixes bug #526247.
+2010-03-08 14:46:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-03 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.
- * configure.ac:
- Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
+2010-03-08 12:11:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basedepay: clarify some documentation
- * 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.
+2010-03-08 11:25:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/alsa/gstalsasrc.c:
+ alsasrc: return right number of bytes that we wrote
- * ext/ogg/gstoggmux.c:
- Fix sample pipeline description.
+2010-03-08 11:20:51 +0100 Dake Gu <gudake@gmail.com>
-2008-04-03 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: fix handling of x-server-ip-address
+ Fix handling of x-server-ip-address.
- * 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
+2010-03-02 11:25:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/design/draft-keyframe-force.txt:
+ docs: update keyframe force event
+ Add field to send all headers.
+
+=== release 0.10.27 ===
+2010-03-06 00:09:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * 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-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-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
- Update introspection data.
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.27
- * ext/ogg/gstoggmux.c:
- Document oggmux.
+2010-03-06 00:08:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin2.c:
- Don't use gtk-doc style comment start for private stuff, but make it
- formatted like this for consistency.
+ * 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
-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>
+2010-03-05 15:58:44 +0000 Tim-Philipp Müller <tim.muller@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.
+ * 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.
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-03 20:18:16 +0000 Tim-Philipp Müller <tim.muller@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.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.4 pre-release
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-03 20:17:31 +0000 Tim-Philipp Müller <tim.muller@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.
+ * po/ja.po:
+ po: update translations
-2008-04-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-03 20:15:44 +0000 Josep Torra Valles <n770galaxy@gmail.com>
- * 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.
+ * 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.
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-01 12:17:40 +0100 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.
+ * 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
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-24 01:10:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
+ * configure.ac:
+ * po/nl.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.3 pre-release
- * 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.
+2010-02-23 16:57:53 +0100 Götz Waschk <waschk@mandriva.org>
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/examples/overlay/Makefile.am:
+ examples: Dist header file for the Qt graphics view example
+ Fixes bug #610832.
- * 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...
+2010-02-23 11:41:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * 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.
+2010-02-19 18:24:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-04-01 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- * configure.ac:
- Require GLib 2.12 and liboil 0.3.14.
+2010-01-27 15:07:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_process_double):
- Unconditionally use liboil 0.3.14 function.
+ * 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
-2008-03-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-21 19:52:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- ms-gsm can have arbitrarty sample rates. See #481354.
+ * 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.
-2008-03-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-22 13:01:19 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- MP4S is generic MPEG-4, not a microsoft variant.
+ * 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.
-2008-03-27 Michael Smith <msmith@fluendo.com>
+2010-02-19 14:40:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Check the body CRC (if set) when depayloading.
- Fixes #522401.
+ * tests/examples/overlay/.gitignore:
+ gitignore: ignore files in new example directroy
-2008-03-24 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-17 14:59:33 +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/video/Makefile.am:
+ make: fix copy and paste error in git rules (audio<->video)
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-19 17:44:18 +0100 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_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.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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.
-
- * 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.
-
-2008-03-23 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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.
-
-2008-03-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/elements/gdpdepay.c:
- Make test compile without unused function/variable warnings on PPC.
-
-2008-03-22 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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>
-
- * 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 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- Back to development - 0.10.18.1
-
-=== release 0.10.18 ===
-
-2008-03-20 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- releasing 0.10.18, "I will follow"
+ * 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.
-2008-03-18 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-19 00:22:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
+ * win32/common/_stdint.h:
* win32/common/config.h:
- 0.10.17.4 pre-release
-
-2008-03-18 Wim Taymans <wim.taymans@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.
-
-2008-03-18 Wim Taymans <wim.taymans@collabora.co.uk>
-
- 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 Sebastian Dröge <slomo@circular-chaos.org>
-
- 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.
-
-2008-03-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
+ 0.10.26.2 pre-release
- * 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.
+2010-02-19 00:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-03-14 David Schleef <ds@schleef.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/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/videotestsrc/videotestsrc.c: Oops, revert last change
- because -base is in freeze.
+2010-02-19 00:17:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-03-14 David Schleef <ds@schleef.org>
+ * tests/examples/overlay/.gitignore:
+ Ignore new overlay examples
- Patch by: William M. Brack
+2010-02-18 23:47:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
+ * 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.
-2008-03-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-18 11:42:55 -0800 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-03-12 Michael Smith <msmith@fluendo.com>
+2010-02-18 12:42:53 +0000 Patrick Radizi <patrick.radizi@axis.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-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: make sure not to dereference NULL username or password
+ Fixes #610268.
-2008-03-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-17 21:22:54 -0800 David Schleef <ds@schleef.org>
- * win32/common/libgstnetbuffer.def:
- Add new symbol gst_netaddress_equal. Fixes bug #521743.
+ * 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.
-2008-03-11 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-16 15:43:26 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
- * win32/common/config.h:
- 0.10.17.3 pre-release
+ * 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>
-2008-03-10 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Fix duration when no clock was provided. Fixes #520300.
+2010-02-15 11:11:04 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- Patch by: Olivier Crete <tester at tester ca>
+2010-02-16 10:09:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * 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).
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-08 13:54:57 +0200 Kaj-Michael Lang <milang@tal.org>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Update mode property docs, it's deprecated now.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Handle stm module format
+ Fixes #609314.
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-15 12:10:10 +0100 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/vorbis/gstivorbisdec.c:
+ ivorbisdec: set rank to SECONDARY
-2008-03-04 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 12:09:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* configure.ac:
- * win32/common/config.h:
- 0.10.17.2 pre-release
-
-2008-03-03 Jan Schmidt <jan.schmidt@sun.com>
-
- * gst/Makefile.am:
- GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
- them twice
+ * ext/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * 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.
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Add new API to the defs
+2010-02-03 14:37:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+ * 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.
- Patch by: Mersad Jelacic <mersad at axis dot com>
+2010-02-15 10:23:13 +0100 Sebastian Dröge <sebastian.droege@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)
+ * tests/check/elements/playbin.c:
+ playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 09:04:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/mixer.c:
- Add a few simple checks for the new message types.
+ * ext/ogg/gstoggparse.c:
+ oggparse: Fix another format string compiler warning
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 08:56:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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:
+ oggdemux: Fix format string compiler warnings
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 08:48:58 +0100 Sebastian Dröge <sebastian.droege@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/playback/gstplaybin2.c:
+ playbin2: Post a missing element message and an error message if no uridecodebin can be found
-2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+2010-02-15 08:46:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
- plug-ins are included/excluded. (#498222)
+ * 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.
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-15 08:28:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for IMelody files, using audio/x-imelody.
- See bug #519516.
+ * tests/check/elements/playbin2.c:
+ playbin2: Enable all unit tests
+ They're all working and valgrind clean now.
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-15 08:26:05 +0100 Sebastian Dröge <sebastian.droege@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.
+ * 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.
- * win32/common/libgstrtp.def:
- Add gst_rtp_buffer_set_extension_data to the symbol definition file.
+2010-02-15 08:20:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
+ and not CORE MISSING_PLUGIN.
- Patch by: José Alburquerque <jaalburqu svn gnome org>
+2010-02-15 08:18:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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).
-
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Don't include unix headers unconditionally (fixes #518037).
-
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
+ playbin2: Free the subtitle URI
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 08:06:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 06:50:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/check/elements/.gitignore:
+ tests: Add decodebin2 test to .gitignore
-2008-02-29 Rene Stadler <mail@renestadler.de>
+2010-02-15 01:18:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/v4l/gstv4lelement.c:
- Add missing semicolon to fix indentation.
+ * 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.
-2008-02-29 Julien Moutte <julien@fluendo.com>
+2010-02-15 01:17:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-02-29 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 00:11:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * ext/ogg/gstoggstream.c:
+ ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
+ See #609252.
-2008-02-29 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-14 23:16:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 96dc793 to 44ecce7
-2008-02-29 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-14 23:10:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
- Enable vis setting.
+ * 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/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.
+2010-02-14 22:16:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-28 David Schleef <ds@schleef.org>
+ * tests/check/elements/playbin.c:
+ playbin: Fix indention in the unit test
- * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
+2010-02-13 01:08:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/volume/gstvolume.c:
+ volume: Replace this variables by self
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2010-02-12 19:43:13 +0100 Josep Torra Valles <n770galaxy@gmail.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.
+ * 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.
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-12 18:00:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix Since tag
- * 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.
+2010-02-12 14:19:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/riff/riff-read.c:
+ riff: treat JUNQ chunks like JUNK chunks
- * 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.
+2010-02-12 14:29:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update basesrc segment duration and post duration messages from the streaming thread
- * tests/check/libs/rtp.c: (GST_START_TEST):
- Add check for RTP buffer defaults, padding and marker bit API.
+2010-02-11 14:10:02 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-27 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/tag/tags.c:
+ tags: improve docs about determining the encoding
- * 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.
+2010-02-11 14:09:05 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ comment: fix wrong header comment
- * 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().
+2010-02-01 13:50:14 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * 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.
+2010-02-01 19:01:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for the smoke video codec. Copied from the jpeg plugin.
+2010-02-01 18:56:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: cleanups
+ Avoid some typechecks.
+ Avoid dereferencing appsrc->priv all the time.
- * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
- (plugin_init):
- Add midi typefinder, copied from the timidity plugin.
+2010-02-01 18:55:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: cleanups
+ Avoid some typecasting.
+ Avoid dereferencing appsink->priv all the time.
- Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
+2010-02-01 15:09:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: avoid some typecasts
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-29 16:34:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-10 16:05:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Brian Cameron <brian dot cameron at sun dot com>
+ * 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.
- * configure.ac:
- Don't hardcode -Wall and -Werror for configure checks, this fails
- with non-GCC compilers. Fixes bug #517991.
+2010-02-10 10:12:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back
- * 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.
+2010-02-09 17:39:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update segment duration and post a duration message if the duration changes
+ Fixes bug #609423.
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
+2010-02-11 10:56:17 +0100 Benjamin Otte <otte@redhat.com>
-2008-02-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/examples/seek/Makefile.am:
+ build: link to libm in examples that use it
+ This fixes build failure in Fedora 13.
- * 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.
+2010-02-11 01:11:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * MAINTAINERS:
+ Update MAINTAINERS, add myself
- * 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.
+2010-02-11 23:57:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure: back to development
+ Slushy freeze remains in effect.
- * 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'.
+=== release 0.10.26 ===
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-10 20:17:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-10 20:16:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- Fix typo in docs blurb.
+ * 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
-2008-02-19 Julien Moutte <julien@fluendo.com>
+2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * ext/theora/gsttheoradec.c:
+ theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
+ Fixes #609252.
- * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
- typefind lookup to fix typefinding on HD clips.
+2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: use the default granpos functions for kate streams
+ Set timestamps on kate packets. See bug #600929.
- * 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).
+2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-19 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.25.3 pre-release
- * 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.
+2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * po/bg.po:
+ po: update translations
- * 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.
+2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-19 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Josep Torra Valles <josep at fluendo dot com>
+2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
- (plugin_init):
- Add typefinding support for h264 elementary streams.
- Fixes bug #517420.
+ * 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.
-2008-02-18 Stefan Kost <ensonic@users.sf.net>
+2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Require CVS of core for new API in collectpads.
+ * 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
- * gst/adder/gstadder.c:
- Use new API to make adder sparse stream aware.
+2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-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.
+ * 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.
- * gst/playback/gststreamselector.c: (gst_selector_pad_event),
- (gst_selector_pad_chain):
- * gst/playback/gststreamselector.h:
- Make sure we forward the event only once.
+2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
- Add and implement the mute button for playbin2.
+ * common:
+ Automatic update of common submodule
+ From 15d47a6 to 96dc793
-2008-02-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * 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
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
- Add some more debug info.
- Make sure we never return a negative delay. Fixes #516246.
+2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ 0.10.25.2 pre-release
- * 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.
+2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Julien Moutte <julien@fluendo.com>
+ * 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
- * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
- some seek flags when changing rate.
+2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer):
- Fix potential leaks.
+2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
- Fix leak when there is no function configured.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
- (gst_v4lsrc_buffer_finalize):
- Correctly chain up the finalize method.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't skip an element when getting the topology
+ Fixes #608167
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
- * ext/gio/gstgiostreamsink.c:
- * ext/gio/gstgiostreamsrc.c:
- Add documentation and example code for giostreamsink/giostreamsrc.
+ * 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.
- * 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.
+2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: properly set up the media type for kate streams
+ See #600929.
- * ext/gio/gstgiosink.c:
- * ext/gio/gstgiosrc.c:
- Make the documentation of giosink/giosrc complete, large parts
- are based on the gnomevfssink/gnomevfssrc docs.
+2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * 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.
+2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@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)
+ * 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
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@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).
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
+
+2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * 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.
-2008-02-11 Julien Moutte <julien@fluendo.com>
-
- Patch by: Alan Peevers <peeves@pacbell.net>
+2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
- * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
- lock when calling alsa methods.
+ * 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.
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@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).
+ typefindfunctions: Add typefinder for ISO MP4 files
+ Fixes bug #607848.
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-24 13:29:07 +0000 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.
+ * 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
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Branko Čibej <brane at xbc dot nu>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Strip trailing \0 for subtitle OGM streams
+ Fixes bug #607870.
- * sys/xvimage/xvimagesink.c:
- Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
- Fixes bug #515654.
+2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-09 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Correctly set DELTA_UNIT flag for OGM streams
- * 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.
+2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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/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.
+2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Extract tags from OGM text streams and don't push them downstream
- * 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>
+2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Revert last change as we now check in gtk-doc.m4 for sed.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
-2008-02-08 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Find and subst SED when building the docs.
+ * 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.
-2008-02-08 Julien Moutte <julien@fluendo.com>
+2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/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.
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- Make DISABLE_DEPRECATED defined *only* during CVS, not during
- pre-releases or releases.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for Zip Block Motion Video
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- * ext/gio/Makefile.am:
- Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
- reporting
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Zip Block Motion Video
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
- * docs/plugins/Makefile.am:
- Add the headers which need scanning for the GIO plugin. The rest of
- the docs still need migrating.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: YUNV is a fourcc which is also used for YUY2 raw video
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:13:45 +0100 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/riff/riff-media.c:
+ riff: vp61 and VP61 are also valid On2 VP6 fourcc
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:10:45 +0100 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/riff/riff-media.c:
+ riff: Add mapping for On2 VP5
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-23 15:04:35 +0100 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-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.
-2008-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix pbutils header.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for LOCO Lossless codec
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Fix compiler warning.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for LOCO Lossless codec
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
- Clear the addrinfo struct using memset. Fixes #514937.
+2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-02-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Autodesk Animator codec
- * gst/tcp/gstfdset.h:
- Remove unused field to same some memory.
+2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Mark action signals as such.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Autodesk Animator Codec
-2008-02-06 Michael Smith <msmith@fluendo.com>
+2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (_theora_granule_frame),
- (_inc_granulepos):
- Increment granulepos for new-bitstream versions appropriately.
- Fixes #514623.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: ...and set caps on queued packet buffers too
-2008-02-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Set caps on header buffers
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@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).
+ * 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
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/typefind/gsttypefindfunctions.c:
+ typefind: include stdio.h for sscanf
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-22 01:49:38 +0000 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.
+ * 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.
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
- * 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.
+ * 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
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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/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>
+2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@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:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
- First round of plugin docs cleansups.
+ * 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.
- * 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.
+2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/Makefile.am:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- Add header for oggmux. the c-file needs a doc blob still.
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoid division by 0 on bad annodex streams
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for y4m container
- * 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()
+2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: ptime/maxptime should be unsigned
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
- * configure.ac:
- Back to CVS
+2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-=== release 0.10.17 ===
+ * 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
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- releasing 0.10.17, "Peanut Butter and Jelly"
+ * common:
+ Automatic update of common submodule
+ From 14cec89 to 15d47a6
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
- * 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
+ * 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.
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Revert patch to gstrtspconnection.c for brown paper bag
- release of -base. Re-opens: #511825
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Reject empty caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=607353
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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
+ * 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
-2008-01-30 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * win32/common/libgstpbutils.def:
- Export the two new _get_type() functions which are needed
- by the python bindings.
+ * 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.
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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.
+ * 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
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for MXF container format
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- Back to CVS
+ * gst/playback/gstplaysink.c:
+ playsink: re-use iterator callback to avoid code duplication
-=== release 0.10.16 ===
+2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+ * 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).
- * configure.ac:
- releasing 0.10.16, "Scheduled Interruption"
+2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-22 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_get_extension_data):
- Fix typos and wrong extension check. Fixes #511274.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: sprinkle some more locking
+ ... to avoid races and ensure some data structure consistency.
+ See also #574289.
-2008-01-18 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * po/sk.po:
- Oops - add new sk.po mentioned in the LINGUAS I just committed
+ * 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.
-2008-01-17 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/LINGUAS:
- Add ca translation to the disted list.
+ * 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.
+
+2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * win32/vs6/libgstsdp.dsp:
- Convert line endings to CRLF
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix pad ref leak
-2008-01-17 Sebastien Moutte <sebastien@moutte.net>
+2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * win32/MANIFEST:
- Add win32/vs6/libgstrtsp.dsp to MANIFEST
+ * docs/plugins/Makefile.am:
+ docs: fix out-of-source build
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * win32/common/libgstsdp.def:
- * win32/common/libgstvideo.def:
- Add new API declarations
+ * tests/icles/stress-playbin.c:
+ stress-playbin: fix error return check
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/gsttheoradec.h:
- * ext/theora/gsttheoraparse.h:
+ * 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:
- Take a 2nd stab at handling libtheora granulepos changes in the decoder
- and parser by inspecting the bitstream version of the incoming data.
+ theora: Rename source files to have the same name as the headers
-2008-01-14 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@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.
+ * 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
-2008-01-13 David Schleef <ds@schleef.org>
+2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videoscale/vs_4tap.c:
- Fix valgrind error on 4tap scaling method.
+ * 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.
-2008-01-13 Sebastien Moutte <sebastien@moutte.net>
+2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
-2008-01-13 Julien Moutte <julien@fluendo.com>
+2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.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.
+ * 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.
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: mp4 video is not parsed
- * 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.
+2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add aac stream-format to caps
+ Also add the aac stream-format field on the caps when
+ detecting it.
- * 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.
+2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst/playback/gstplaysink.c:
+ playsink: Fix handling of the native audio/video flags
+ Fixes bug #606687.
- * gst-libs/gst/tag/Makefile.am:
- Fix include path order
+2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-11 Edward Hervey <edward.hervey@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix unitialized variable.
+ If the package isn't handled, gracefully return GST_FLOW_OK.
- * 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.
+2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-11 Michael Smith <msmith@fluendo.com>
+ * 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).
- * 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
+2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ docs: minor netbuffer documentation fix
- * 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.
+2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-10 Sebastian Dröge <slomo@circular-chaos.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/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.
- * 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.
+2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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).
+2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description strings for ID3, APE, and ICY tags.
+2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-plugins-base.spec.in:
+ Add missing source file for tagger to Makefile and update spec file
- * 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>
+2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
- Patch by: Bastien Nocera <hadess at hadess net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: handle 32 bit raw RGB video.
- * 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).
+2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-01-07 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: decide flac header packet by content rather than count
- Patch by: Jerone Young <jerone at gmail com>
+2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: reset header packet count at bos page
-2008-01-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: add support for buffer-lists
-2008-01-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2008-01-05 Julien Moutte <julien@fluendo.com>
+2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
- debugging information to understand how X calculates the stride
- for XvImages.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Ignore zero framerate
+ https://bugzilla.gnome.org/show_bug.cgi?id=606163
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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.
+ * 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
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
- Don't set element details for the abstract GstAudioFilter class.
+ * 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
-2008-01-02 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
- Implement get_unit_size() vmethod of GstBaseTransform.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Accept maxptime from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
-2008-01-01 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: enhance flac packet duration calculation
-2007-12-31 David Schleef <ds@schleef.org>
+2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/pipelines/theoraenc.c:
- Reenable theoraenc test, which fails on the buildbot but
- not locally.
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2007-12-31 David Schleef <ds@schleef.org>
+2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/.cvsignore:
- * docs/plugins/.cvsignore:
- Add *-undeclared.txt to fix buildbot.
+ * 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.
-2007-12-31 David Schleef <ds@schleef.org>
+2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Second attempt at disabling theoraenc test long enough to
- get buildbot to compile -base.
+ * 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.
-2007-12-31 David Schleef <ds@schleef.org>
+2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/pipelines/theoraenc.c:
- Disable theoraenc test long enough to get the buildbot to
- compile a recent -base.
+ * 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.
-2007-12-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2007-12-31 Thijs Vermeir <thijsvermeir@gmail.com>
+2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * sys/ximage/ximagesink.c:
- fix typo
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: correctly eat empty and dummy buffers
-2007-12-30 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/gstssaparse.c:
- combine if's
+ * 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
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/adder/gstadder.c:
+ adder: fix juvenile comment
- * gst/playback/test7.c: (about_to_finish_cb):
- Fix test.
+2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix typo in debug message
- * 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.
+2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
- (test_basetransform_based):
- Some test pipelines for the above (not entirely valgrind clean yet
- apparently).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some type checks
-2007-12-21 David Schleef <ds@schleef.org>
+2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Add additional RGBA and RGB-24 video formats.
+ * gst/playback/gstplaybin2.c:
+ playbin2: avoid leaking selector request pads
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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.
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
- * gst/playback/gststreamselector.c: (gst_selector_pad_event):
- Don't leak event.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: don't name the queue
+ There is no reason to name the queue.
+ Fixes #605219
-2007-12-20 Thijs Vermeir <thijsvermeir@gmail.com>
+2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-read.c:
- Use GST_ROUND_UP_2 macro
+ * win32/common/libgstrtp.def:
+ defs: update defs with new symbols
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/.cvsignore:
- Ignore more.
+ * 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
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 47cb23a to 14cec89
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gststreamselector.c: (gst_selector_pad_init):
- Init the pad segment too.
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: add Since marker for the new tolerance property
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/tag/lang.c:
+ docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
- * 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.
+2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-18 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:
+ tests: don't use deprecated GLib API g_mapped_file_free
+ Fixes #605100.
- * gst-libs/gst/interfaces/interfaces.h:
- Also remove interfaces.h from CVS as it is not needed anymore.
+2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * gst-libs/gst/interfaces/Makefile.am:
- interfaces.h is not used anymore so remove it from the build
- process.
+2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-17 David Schleef <ds@schleef.org>
+ * 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.
- * 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.
+2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-17 David Schleef <ds@schleef.org>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add some debugging
- * 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.
+2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * Makefile.am:
- Use new common/win32.mak.
+2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-17 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * 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.
+2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/audiorate/Makefile.am:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ audiorate: use separate header file
- * 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.
+2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: set DISCONT when resyncing (e.g. newsegment)
- * 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).
+2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: also fill up segments if possible
- * 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.
+2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-15 Tim-Philipp Müller <tim at centricular dot net>
+ * 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-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).
+2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-12-14 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: properly report truncated samples as dropped samples
- * 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.
+2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/tag/lang.c:
+ docs: mention that gst_tag_get_language_name() may return NULL
- * 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.
+2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/libs/tag.c:
+ checks: some more testing for the new language code functions
- * 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.
+2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ docs: misc. mixer docs improvements
- * 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.
+2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ docs: add short descriptions for API reference contents page
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_free):
- Close control sockets. Fixes #503440.
+ * gst-libs/gst/tag/lang-tables.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ tag: make internal language names table static
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
- Expose the right pad in the right place with the right element.
+ * 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.
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description for 'private' dts caps (who come up with that name?).
+ * .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()
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * Makefile.am:
- Add check-exports target and run it with 'make check'.
+ * ext/ogg/gstoggstream.c:
+ ogg: ogm video has constant packet duration
- * configure.ac:
- Be stricter about what we export in our libraries: change regexp so that
- we only export _gst_foo(), but not __gst_foo().
+2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
- * 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.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: implement old fLaC mapping
- * win32/common/libgstaudio.def:
- Add missing symbols.
+2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-11 David Schleef <ds@schleef.org>
+ * 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
- * ChangeLog: remove conflict markers
+2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-11 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: remove redundant fields
- * 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.
+2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-10 Wim Taymans <wim.taymans@gmail.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/audioconvert/Makefile.am:
- Also link to libm.
+2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-10 Wim Taymans <wim.taymans@gmail.com>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: fix peer query
- * 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.
+2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-10 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoradec.c:
+ theoradec: fix query
- * 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.
+2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/theora/theoradec.c:
+ theoradec: small cleanups
- * configure.ac:
- Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
+2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-08 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: use gst_pad_peer_query()
- Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
+2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
- * tests/check/elements/subparse.c: (srt_input), (srt_input0),
- (test_src):
- Add unit test for the above.
+2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-06 Wim Taymans <wim.taymans@gmail.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only flush downstream on seeks for flushing seeks
- * gst/playback/gstplay-enum.c:
- (register_gst_autoplug_select_result),
- (gst_autoplug_select_result_get_type):
- * gst/playback/gstplay-enum.h:
- Add missing files.
+2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-05 Wim Taymans <wim.taymans@gmail.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
- * 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.
+2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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/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.
+2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-03 Edward Hervey <bilboed@bilboed.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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
+2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: improve keyframe seeking
+ Improve keyframe seeking.
+ Fix reverse playback.
- * 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
+2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-11-30 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/playback/gstplaysink.c:
+ playsink: Some minor cleanup
-2007-11-26 Stefan Kost <ensonic@users.sf.net>
+2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/volume/gstvolume.c:
- * gst/volume/gstvolume.h:
- Add GAP-flag support.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
-2007-11-24 Julien MOUTTE <julien@moutte.net>
+2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-2007-11-23 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (audioresample_query), (audioresample_query_type),
- (gst_audioresample_set_property):
- Implement latency query.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add comment
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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()
+ * 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.
- * 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()
+2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-11-20 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- Patch by: Joe Peterson <lavajoe at gentoo dot org>
+2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix compilation on FreeBSD (Gentoo). Fixes #498228.
+ * 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.
-2007-11-19 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
- * tests/check/libs/fft.c: (GST_START_TEST):
- Fix scaling to really have dB instead of something else.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: fix for I-frame-only theora
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2009-12-03 01:16:17 +0000 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.
+ * ext/ogg/gstoggstream.c:
+ ogg: log when ogg mapper doesn't accept the setup header packet
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): Try to support stable version
- of GTK.
+ * ext/ogg/gstoggstream.c:
+ ogg: extract width, height and PAR from theora header and add to caps
-2007-11-17 Stefan Kost <ensonic@users.sf.net>
+2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/README:
- * gst/playback/test7.c:
- Fix the build + little README update.
+ * ext/ogg/gstoggstream.c:
+ ogg: extract number of channels from FLAC, speex and vorbis headers
+ Because we can.
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
- Add playbin2 seek pipeline.
+ * gst/playback/gstplaybin2.c:
+ build: fix build with debug logging disabled.
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@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.
+ * 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’
- * 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).
+2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * gst-plugins-base.doap:
- Oops, fix the release name.
+2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * gst-plugins-base.doap:
- Add 0.10.15 release
+2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/videoscale/vs_4tap.c:
+ Build fix for MSVC
- * configure.ac:
- Back to CVS
+2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
-=== release 0.10.15 ===
+ * gst/subparse/qttextparse.c:
+ build: add missing includes for sprintf and atoi
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * configure.ac:
- releasing 0.10.15, "No need to argue"
+ * 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
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * win32/vs6/libgstfft.dsp:
- Convert line endings to DOS.
+ * 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>
-2007-11-15 Sebastien Moutte <sebastien@moutte.net>
+ * gst/subparse/gstsubparse.c:
+ subparse: conditionally cleanup sami context
+ Only cleanup sami context if we are parsing sami subtitles,
+ otherwise we might have crashes.
- * 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
+2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+ * 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
- * win32/vs6/libgstinterfaces.dsp:
- * win32/vs6/libgstrtsp.dsp:
- Convert line endings back to DOS
+2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+ * common:
+ Automatic update of common submodule
+ From 87bf428 to 47cb23a
- * 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
+2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-11-09 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ Automatic update of common submodule
+ From da4c75c to 87bf428
- * gst/playback/gstdecodebin2.c:
- Dont leak ghostpad. Fixes #475451.
+2009-11-30 10:22:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-11-09 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Fix some pad refcount issues
+ Fixes bug #603345.
- * docs/design/design-decodebin.txt:
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- Update some more docs and comments.
+2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 53a2485 to da4c75c
- Patch by: Sebastien Moutte <sebastien moutte net>
+2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
- * 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).
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: handle theora streams with 0 keyoffset
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstaudio.dsp:
- * win32/vs6/libgstrtp.dsp:
- * win32/vs6/libgsttag.dsp:
- Update vs6 projects files (#494346).
+2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle unknown streams
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+2009-11-26 14:30:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * 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.
-2007-11-06 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
+ ...and fix code style a bit.
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
- (finish_source):
- Avoid crash when there are external subtitles (fixes #491722).
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ audiofilter: Add _CAST variants of the cast macros
-2007-11-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-25 10:26:16 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
+ Fixes bug #602834 and #350748.
- * win32/common/libgstrtp.def:
- Add some more missing symbols (#492813).
+2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
- * 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.
+ * 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.
-2007-11-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+ * 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>
- * 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/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.
- * 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).
+2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
- * gst-libs/gst/sdp/Makefile.am:
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix includes and LIBS for win32/Mingw (#492306).
+ * 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.
- * tests/examples/dynamic/addstream.c (pause_play_stream):
- Use more portable g_usleep() instead of sleep() (#492306).
+2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
-2007-11-01 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: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * ext/ogg/gstoggmux.c:
+ oggmux: don't overwrite object properties
- * 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).
+2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstringbuffer.h:
- No trailing commas in enum list (for gcc-2.9x).
+ * 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.
- * 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).
+2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
- * win32/common/config.h.in:
- Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ debug: fix format string that was missing a var
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Export a few more symbols (#492114).
+2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-11-01 Stefan Kost <ensonic@users.sf.net>
+ * gst/adder/gstadder.c:
+ * tests/check/elements/adder.c:
+ adder: make events succeed, if they succed on atleast one pad
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- Readd the deprecation guards, but preserve compilability.
+2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix stupid bug introduced in last commit
- * tests/check/elements/decodebin.c: (test_text_plain_streams):
- Make sure the pipeline really operates in push mode as it should
- in this case.
+2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-31 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.
- * 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.
+2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-31 Stefan Kost <ensonic@users.sf.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.
- * 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.
+2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 0702fe1 to 53a2485
- * 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.
+2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-10-30 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * 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.
+2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-10-30 Stefan Kost <ensonic@users.sf.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.
- * tests/icles/stress-playbin.c:
- Fix the build.
+2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-30 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-11-15 19:36:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+ * 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
- * 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.
+2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-10-28 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix 'uninitialized' compiler warning
- * 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.
+2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-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).
+ * configure.ac:
+ configure: bump core requirement to 0.10.25.1
+ We depend on new API that's only in git so far.
- * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
- Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
+2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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-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
-2007-10-24 Wim Taymans <wim.taymans@gmail.com>
+2009-11-17 16:39:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Inspired by patch of: René Stadler <mail at renestadler dot de>
+ * 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.
- * 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.
+2009-11-17 16:37:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-10-23 Tim-Philipp Müller <tim at centricular dot 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-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).
+2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-10-22 Wim Taymans <wim.taymans@gmail.com>
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: remove unused array
- * 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.
+2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: Use new double->fraction transformation function from core
- * 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.
+2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
- * 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.
+2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
- * 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.
+2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't leak the GError and debug string when parsing error messages
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/stress-playbin.c:
- Does what it says on the tin.
+2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
-2007-10-15 Wim Taymans <wim.taymans@gmail.com>
+ * 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
- * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
- Fix queue negotiation. See #486758.
+2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * 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.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
- (gst_xvimagesink_xwindow_new),
- (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
+2009-11-11 17:08:47 +0100 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.
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Only recreate factory caps if necessary and cache them
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-10 18:27:15 +0100 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/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.
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Jason Kivlighn <jkivlighn gmail com>
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Use gst_pad_get_caps_reffed()
- * gst-libs/gst/tag/gstid3tag.c:
- * tests/check/libs/tag.c:
- Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
+2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-10 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain):
- Use new basesink method to make our EOS drain interruptable.
+2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-10 Jan Schmidt <Jan.Schmidt@sun.com>
+ * 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.
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Fix silly search-replace oversight.
+2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-09 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Set the video framerate on parsers if possible
+ Fixes bug #599649.
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
- (gst_basertppayload_set_outcaps):
- Fix caps memleak. Fixes #484989.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Make fps a GstFraction typed property and use it properly
+2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Add property for the video framerate
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
- Fix debug output.
+2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-08 Jan Schmidt <Jan.Schmidt@sun.com>
+ * 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.
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
- to avoid compiler warnings
+2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Add a queue before subtitleoverlay
+ This will improve playback, and the same thing is done
+ for subpicture streams too.
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't leak caps.
+2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: Use subtitleoverlay for subtitles
- * 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.
+2009-11-02 07:43:42 +0100 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.
+ * 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
-2007-10-05 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-04 Stefan Kost <ensonic@users.sf.net>
+2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Replace g_return_if_val (as it could be disabled), with regular return
- and warning.
+ * ext/theora/theoradec.c:
+ theoradec: Keep timestamp from incoming buffer if it is valid
+ Fixes bug #601627.
-2007-10-03 Stefan Kost <ensonic@users.sf.net>
+2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/simple-launch-lines.c:
- Print message name and not just number.
+ * 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.
-2007-10-02 Wim Taymans <wim.taymans@gmail.com>
+2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-02 Stefan Kost <ensonic@users.sf.net>
+2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
- * tests/examples/snapshot/snapshot.c:
- Fix the build.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clear the EOS state on a seek.
+ Allow seeking back into the stream after it hits EOS.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Add RFC 3556 bandwidth modifiers.
+ * 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
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-10 00:48:42 +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:
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Update documentation.
+ * gst/playback/gstplaysink.c:
+ playsink: assign chain->mute before using it
+ Fixes GObject warnings when starting totem.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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-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.
+2009-11-09 19:58:20 +0100 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.
+ * gst/playback/gstfactorylists.c:
+ playback: Fix the order in strcmp that I broke in previous commit.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-09 19:16:21 +0100 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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Reduce number of calls to gst_type_find_peek.
+ Shaves off a couple percents off typefinding
-2007-09-29 Sebastien Moutte <sebastien@moutte.net>
+2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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/playback/gstfactorylists.c:
+ playback: Avoid expensive API calls in tight loop.
+ We know we're dealing with GstPluginFeature.
- * tests/examples/snapshot/snapshot.c: (main):
- Print error when pipeline failed to construct.
+2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Add unit test for property settings
+ Also includes a regression test for bug #601104.
- * 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.
+2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Never return a negative track number in get_uri()
- * 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>
+2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@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/cdda/gstcddabasesrc.c:
+ cddabasesrc: Don't set the track to 1 every time a device is set
+ Fixes bug #601104.
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/hu.po:
- * po/sv.po:
- * po/uk.po:
- Updated translations to 0.10.14
+ * gst/playback/gstinputselector.c:
+ inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- translated by: Jakub Bogusz <qboosh@pld-linux.org>
+ * 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.
- * po/pl.po:
- Added Polish translation.
+2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- translated by: Ilkka Tuohela <hile@iki.fi>
+2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/fi.po:
- Added Finnish translation.
+ * gst/playback/gstinputselector.c:
+ inputselector: Make sure that running_time->timestamp calculation never becomes negative
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
- translated by: Jorge González González <aloriel@gmail.com>
+ * 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.
- * po/es.po:
- Added Spanish translation.
+2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Improve all-raw-caps detection for pads
- translated by: Mogens Jaeger <mogens@jaeger.tf>
+2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * po/da.po:
- Added Danish translation.
+ * 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
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- translated by: Funda Wang <fundawang@linux.net.cn>
+ * 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.
- * po/zh_CN.po:
- Added Chinese (simplified) translation.
+2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Synchronize video/text based on the running time
+ Instead of simply using the buffer timestamps.
- translated by: Alexander Shopov <ash@contact.bg>
+2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/bg.po:
- Added Bulgarian translation.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Clip text buffers to the text segment and reset segments properly
-2007-09-21 Wim Taymans <wim.taymans@gmail.com>
+2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_push_one):
- Fix compilation wrt printf arguments.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Put the video segment into the instance struct instead of allocating it separately
-2007-09-20 Wim Taymans <wim.taymans@gmail.com>
+2009-11-06 09:05:09 +0100 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.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Check if text timestamp/duration is valid before clipping
-2007-09-20 Stefan Kost <ensonic@users.sf.net>
+2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/generic/states.c:
- Improved state change unit test.
+ * ext/theora/theoradec.c:
+ theoradec: printf format fix
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
- * docs/plugins/.cvsignore:
- * tests/check/.cvsignore:
- Ignore registries in any format.
+ * gst/gdp/gstgdpdepay.c:
+ gdpdepay: Clear adapter on flush and state change
+ Fixes #600469
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gstinputselector.c:
+ inputselector: use _get_caps_reffed()
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c:
- Improve debug logs a bit and be more verbose if things go wrong.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2: avoid copying caps
+ Use get_caps_refed() when we can.
-2007-09-17 Jan Schmidt <Jan.Schmidt@sun.com>
+2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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/audiorate/gstaudiorate.c:
- Always pull in config.h before including any system headers.
+2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-17 Wim Taymans <wim.taymans@gmail.com>
-
- * 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 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/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.
- * 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.
+2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-15 Wim Taymans <wim.taymans@gmail.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-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
- Add some debug info when negotiating caps.
+2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-14 Wim Taymans <wim.taymans@gmail.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.
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- A buffer with an empty payload is also a valid buffer.
+2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstinputselector.c:
+ inputselector: also add inline to the proto to fix the build
+ Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
- * 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.
+2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Initialize caps property with the default raw caps
- * 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.
+2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-14 Julien MOUTTE <julien@moutte.net>
+ * 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.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
- typefind for QCP files (RFC #3625)
+2009-11-03 08:26:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-13 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * 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.
+2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
-2007-09-12 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.
- * 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.
+2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
+ * po/POTFILES.in:
+ po: queue2 has moved to core
- * 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.
+2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-11 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/playback/gstplaysink.c:
+ playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
-2007-09-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
+ Fixes bug #600027.
- * 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).
+2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-10 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
- * 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
+2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-10 Stefan Kost <ensonic@users.sf.net>
+ * docs/plugins/inspect/plugin-queue2.xml:
+ queue2: Remove inspect file
- * gst/playback/gstdecodebin2.c:
- Don't leak request pads. Fixes #475395.
+2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstqueue2.c:
+ queue2: Remove from gst-plugins-base
+ This is now in coreplugins.
- Patch by: René Stadler <mail at renestadler dot de>
+2009-10-28 11:29:36 +0200 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.
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ docs: include more indexes
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/rtp/gstrtpbuffer.c:
- * tests/check/libs/rtp.c:
- Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
+ rtp: dump packets which we reject
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore basetime unit test binary
- * 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).
+2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-09-07 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.
- * 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.
+2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-06 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstqueue2.c:
+ queue2: add custom acceptcaps function
- * configure.ac:
- Fix configure check for HAVE_LIBXML_HTML.
+2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement low/high watermark property
- * tests/check/libs/.cvsignore:
- Ignore more, in case the build bots work again one day.
+2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/examples/seek/seek.c:
+ seek: add checkbox to enable buffering
- Reviewed by: Stefan Kost <ensonic@users.sf.net>
+2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@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.
+ * 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).
- * 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.
+2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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>
+ * 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-libs/gst/rtp/gstrtpbuffer.c:
- Need to include stdlib.h for abs() here too.
+2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: buffering is implemented now
- * gst/playback/gststreaminfo.c:
- Fix build.
+2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-05 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: buffering is implemented now
- * gst/playback/gststreaminfo.c:
- Clean up some half-disabled code and comment.
+2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-04 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: configure use-buffering on multiqueue
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use 0 for max buffer size
- * 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.
+2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Bump the MTU to 1400.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: set some reasonable defaults
-2007-09-03 Johan Dahlin <jdahlin@async.com.br>
+2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c (plugin_init):
- Add an audio/x-nsf typefind function for the nsfdec element.
+ * 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.
-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
+2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add use-buffering property
+ Add a use-buffering property that will perform buffering on the parsed or
+ demuxed media.
- * 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()
+2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-09-03 Wim Taymans <wim.taymans@gmail.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-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.
+2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- Fix typo.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: move error path down
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+2009-10-23 11:02:40 -0400 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.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement max queue size properties
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+2009-10-23 10:42:23 -0400 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.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add properties for buffering
+ Add properties that can be used to configure the multiqueue buffers and
+ buffering methods
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.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.
+ * 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
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_sdes_copy_entry):
- Use g_strndup which does exactly what we want.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't destroy the suburidecodebin on errors
+ It can still be reused
- * 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()
+2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-30 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
- * 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()
+2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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.
+2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
+ * 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
- Patch by: Daniel Díaz <yosoy@danieldiaz.org>
+2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
- * configure.ac:
- * gst/Makefile.am:
- Check if libxml provides HTML parser which subparse needs.
- Fixes #451970.
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c:
+ theora: Convert theoraparse to libtheora 1.0 API
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
- * ext/alsa/gstalsa.c:
- Fix typo and compilation on big endian systems.
+ * 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
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
- * gst/subparse/gstssaparse.c:
- Convert SSA newline codes into actual newline characters (#470766).
+ * 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.
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
- * 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).
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Removing unused variable
+ The in_stop variable was never read. Removing it.
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
- * 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.
+ * 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
-2007-08-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-08-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_plugin_message_get_installer_detail):
- Add missing separator in PID fallback case.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c:
+ subparse: Add support for DKS subtitle format
+ Fixes bug #598936.
-2007-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@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.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Do shading as first operation
- * gst/audioconvert/Makefile.am:
- Add flags from -base before any more-remote dependencies.
+2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only use a single cairo surface for drawing
+ ... and comment/optimize what is going on here a bit better.
- Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
+2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-2007-08-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/Makefile.am:
- Fix even more.
+ * 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.
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
+2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix typo in previous mxf typefinder change
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
+2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/volume/gstvolume.c:
- Move passthrough below gst_object_sync_values(). Fixes #442654.
+ * 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 :)
-2007-08-22 Wim Taymans <wim.taymans@gmail.com>
+2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- Clarify the docs a little.
+ * gst/playback/gstfactorylists.c:
+ decodebin2: avoid type checks
-2007-08-22 Stefan Kost <ensonic@users.sf.net>
+2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/volume/gstvolume.c:
- Enable liboil for float and add more details about problems with
- int16.
+ * 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.
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
- Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
+ * gst/playback/gstdecodebin2.c:
+ gst/decodebin2: Don't expose chains if we're shutting down.
+ This avoids adding flushing pads to ourself
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@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.
+ * gst-libs/gst/rtsp/.gitignore:
+ .gitignore: update after files got renamed
-2007-08-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@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.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: small comment fix
-2007-08-17 Wim Taymans <wim.taymans@gmail.com>
+2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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/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-libs/gst/rtsp/gstrtspdefs.h:
- Fix compilation with -pedantic.
- Add enum for _poll.
+2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-08-16 Wim Taymans <wim.taymans@gmail.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/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.
- Patch by: Olivier Crete <tester at tester ca>
+2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
-2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
- Only post buffering messages when we are a stream.
+ * common:
+ Automatic update of common submodule
+ From 85d1530 to 0702fe1
-2007-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
- * gst-libs/gst/pbutils/install-plugins.c:
- * gst-libs/gst/pbutils/missing-plugins.c:
- Small docs fix and addition.
+ * ext/theora/theoradec.c:
+ theora: Make theoradec use gstvideo for image conversion
+ Vastly simplifies code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-textoverlay.c:
- Add a dumb little test for textoverlay alignments.
+ * 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
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
- Patch by: Dan Williams <dcbw redhat com>
+ * 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
- * 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.
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
-2007-08-12 Wim Taymans <wim.taymans@gmail.com>
-
- * 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.
+ * 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
-2007-08-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
- * 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)
+ * 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
-2007-08-10 Stefan Kost <ensonic@users.sf.net>
+2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
- * docs/libs/Makefile.am:
- Remove cruft and do some cleanups.
+ * 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
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- Prepare for comming gtkdoc features (rebase against online docs).
+2009-09-09 21:57:08 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-10 Michael Smith <msmith@fluendo.com>
+ * 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
- * 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.
+2009-09-09 21:50:57 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * gst/playback/gstqueue2.c:
- * gst/videorate/gstvideorate.c:
- Printf format fixes (#465028).
+2009-09-09 21:48:08 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-09 Michael Smith <msmith@fluendo.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Disable keyframe_threshold property
+ It's ignored by the current theora encoder
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * 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.
+2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-09 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * gst/playback/gstplaybasebin.c: (group_commit),
- (queue_remove_probe), (queue_threshold_reached):
+2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
- 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.
+ * 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
- Also, fix the spelling of Josep's surname in the ChangeLog.
+2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-09 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * ext/ogg/gstoggmux.c:
- Do not leak oggmux instance.
-
- * ext/vorbis/vorbisenc.c:
- Also log values.
+2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
-2007-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- * po/hu.po:
- * po/it.po:
- * po/nl.po:
- * po/uk.po:
- * po/vi.po:
- Updated translations.
+2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-08-08 Stefan Kost <ensonic@users.sf.net>
+ * 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
- patch by: Yang Hong <hongyang@redflag-linux.com>
+2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextoverlay.h:
- Add 'silent' property to GstTimeOverlay. Fixes #462979
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post a element message on the bus with the stream topology
+ Fixes bug #598533.
-2007-08-08 Wim Taymans <wim.taymans@gmail.com>
+2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Josep Torra Valles <josep@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.
- * 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.
+2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Store for every chain, which pad resulted in its creation
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
- * configure.ac:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect):
- Fix compilation on windows. Fixes #464320.
+ * 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.
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From a3e3ce4 to 85d1530
- * 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.
+2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
- Properly respond to conversion queries. Fixes #464079.
+2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix timestamp comparission, Fixes #597407
- * 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.
+2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Jens Granseuer <jensgr at gmx dot net>
+2009-10-08 19:55:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioquantize.c:
- Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
+ * 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.
-2007-08-03 Wim Taymans <wim.taymans@gmail.com>
+2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
- Add rdt manager for rdt transport.
- Fix parsing of RDT transport.
+ * 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.
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Back to CVS
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: track active conversion in perf log
-=== release 0.10.14 ===
+2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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:
- releasing 0.10.14, "Light Years Ahead"
+2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: whitespace fixes
- * tests/check/libs/audio.c: (GST_START_TEST):
- Fix the test to reflect the behaviour of gst_audio_clip_buffer.
+2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/theoradec.c:
+ theoradec: avoid confusing error
- * gst-libs/gst/audio/audio.c:
- When clipping a buffer with no timestamp, assume it is
- within the segment without warnings.
+2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
- Fixes: #460978
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: fixes warings in macosx snow leopard
-2007-07-27 Wim Taymans <wim.taymans@gmail.com>
+2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
- Fire the signal on the object, not the interface.
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraparse.c:
+ theora: fixes warnings on macosx snow leopard
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst-libs/gst/rtsp/.cvsignore:
- Ber. Don't include the full path, idiot.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ ogg: fixes warnings on macosx snow leopard
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst-libs/gst/rtsp/.cvsignore:
- Ignore generated files.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix a warning in macosx
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ * gst-libs/gst/tag/tags.c:
+ tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * tests/check/gst-plugins-base.supp:
+ check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst-libs/gst/riff/Makefile.am:
- Pull in our dependencies from -base before those from outside.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ audioconvert: change the format instead of cast as ensonic asked
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
- * 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.
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
+ redo of valid part of my previous revert.
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
- * tests/check/libs/.cvsignore:
- Ignore the mixer test binary.
+ * 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.
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.com>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
- Gratuitous comment change to trigger a rebuild on the buildbots.
+ * 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'
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
- * 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.
+ * gst/videorate/gstvideorate.c:
+ videorate: fix warning in macosx
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
- * 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>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: fix warning in macosx
- * 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.
+2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+ * common:
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
- * 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.
+2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ audio: fix warnings building on macosx
- * 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.
+2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: chwck formats just once per _chain()
- * 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.
+2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
- (vorbis_dec_flush_decode):
- Use the new buffer clipping function from gstaudio here.
+2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 19fa4f3 to a3e3ce4
- * 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.
+2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
-2007-07-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Fix type-punning warning
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Cleanup the docs.
+2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-23 Wim Taymans <wim.taymans@gmail.com>
+ * 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
- Patch by: Dan Williams <dcbw at redhat dot com>
+2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/gstdecodebin2.c:
+ decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
-2007-07-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gsturidecodebin.c:
- Init debug category before using it.
+ 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.
-2007-07-21 Jan Schmidt <thaytan@noraisin.net>
+2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
-
-2007-07-20 Jan Schmidt <thaytan@noraisin.net>
+ * 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.
- * 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:
+2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
- Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
- Fixes: #152864
+ * 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.
- Add support for notifying mixer changes on the message bus, and
- implement it in alsamixer.
+2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
- 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
+ * 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.
-2007-07-20 Michael Smith <msmith@fluendo.com>
+2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * tests/check/libs/video.c:
+ tests/check/libs/video.c: Update strides for Y41B
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/examples/volume/Makefile.am:
- * tests/icles/Makefile.am:
- We should use $(LIBM).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: we can use GLib 2.18 API unconditionally now
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/icles/Makefile.am:
- This needs -lm.
+ * configure.ac:
+ configure: bump GLib requirement to 2.18
+ Bump required GLib version as per the release planning docs.
-2007-07-16 Wim Taymans <wim.taymans@gmail.com>
+2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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-libs/gst/interfaces/tuner.c:
+ docs: clarify GstTuner docs in two places
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Bastien Nocera <hadess at hadess dot net>
+ * 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.
- * 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.
+2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstogmparse.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/mpl2parse.c:
+ * gst/subparse/tmplayerparse.c:
+ build: sprintf, sscanf need stdio.h
- Patch by: Jorn Baayen <jorn at openedhand dot com>
+2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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
+ * 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.
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * gst-libs/gst/video/video.h:
+ Update Since tags for NV12/NV21
+ They are added in 0.10.26 now, not 0.10.25
-2007-07-13 Wim Taymans <wim.taymans@gmail.com>
+2009-09-23 15:31:50 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-23 11:03:57 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-23 10:25:02 +0200 Benjamin Otte <otte@gnome.org>
- * 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-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Add NV12 and NV21 formats
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * 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
-2007-07-12 Wim Taymans <wim.taymans@gmail.com>
+2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Fix Y41B
+ Chroma components should be aligned on 4byte boundaries.
+ https://bugzilla.gnome.org/show_bug.cgi?id=595849
-2007-07-12 Wim Taymans <wim.taymans@gmail.com>
+2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * configure.ac:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ streamvolume: Define cbrt() if it's not available
+ Fixes build on Win32, bug #597537.
-2007-07-12 Stefan Kost <ensonic@users.sf.net>
+2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Use pkg-config to locate check.
+ * gst/playback/gstfactorylists.c:
+ factorylist: Use gst_caps_can_intersect() instead of _intersect()
+ This is faster and results in less allocations.
-2007-07-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * tests/check/elements/volume.c: (GST_START_TEST):
- Fix 'make check' build against core CVS.
+ * 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.
-2007-07-10 Stefan Kost <ensonic@users.sf.net>
+2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/interfaces/propertyprobe.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Make gtk-doc happy.
+ * 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.
-2007-07-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Improve debugging of pad blocks
-2007-07-06 Stefan Kost <ensonic@users.sf.net>
+2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix location of includes in the docs.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
-2007-07-06 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
- * 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
+ * configure.ac:
+ configure: Add an 'else' to pangocairo check
+ Otherwise it exits if it fails.
-2007-07-05 Stefan Kost <ensonic@users.sf.net>
+2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Simplify --extra-dir as gtkdoc scans recursively.
+ * 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.
-2007-07-03 Wim Taymans <wim.taymans@gmail.com>
+2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@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.
+ * 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
-2007-06-29 Wim Taymans <wim.taymans@gmail.com>
+2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
- * gst/audioconvert/audioconvert.c:
- Include math.h to fix compilation.
+ * 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
-2007-06-29 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
* 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/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use the ffmpegcolorspace debug category
+ Move gstffmpegcodecmap debug to the ffmpegcolorspace category
-2007-06-28 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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/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.
- * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
- Add unit test for the above.
+2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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/typefind/gsttypefindfunctions.c:
+ typefind: detect 'ftypqt ' as video/quicktime
- * 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.
+2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-06-27 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ back to development -> 0.10.25.1
- * 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.
+=== release 0.10.25 ===
-2007-06-25 Stefan Kost <ensonic@users.sf.net>
+2009-10-05 13:56:15 +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-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-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-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
- Update docs with caps info.
+ * gst-plugins-base.doap:
+ Release 0.10.25
-2007-06-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * po/POTFILES.in:
- Add more files with translatable strings (#450875).
+ * 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
+
+2009-10-01 17:17:55 +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/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
-2007-06-23 Edward Hervey <edward@fluendo.com>
+2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
-2007-06-22 Jan Schmidt <thaytan@noraisin.net>
+2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * MAINTAINERS:
- Updating all the maintainers files
+ * gst/playback/gstplaysink.c:
+ playsink: make the lock recursive for now
+ Fixes #583255
-2007-06-21 Stefan Kost <ensonic@users.sf.net>
+2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@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.
+ * gst/playback/gstplaysink.c:
+ playsink: fix the vis property getter
-2007-06-20 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@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>
+ * gst-plugins-base.spec.in:
+ Add missing file to spec file
-2007-06-19 Andy Wingo <wingo@pobox.com>
+2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_init): Enable pull-mode operation.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
-2007-06-19 Michael Smith <msmith@fluendo.com>
+2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
- * 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-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.
-2007-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-17 01:20:45 +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/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:
- Update translations.
+ * po/zh_CN.po:
+ 0.10.24.3 pre-release
-2007-06-15 David Schleef <ds@schleef.org>
+2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
- * gst/playback/gstqueue2.c:
- Fix compile error from ignored return value.
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: don't ever return NULL in list of strings.
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/playback/gstplaysink.c:
+ playsink: Expose mute,volume,vis-plugin and font-desc properties
+ https://bugzilla.gnome.org/show_bug.cgi?id=594623
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst/playback/gstplaysink.c:
+ GstPlaySink: Expose 'reconfigure' as an action signal.
-2007-06-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 11:17:28 +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/gstplaysink.c:
+ GstPlaySink: Expose flags as a gobject property.
-2007-06-13 Edward Hervey <edward@fluendo.com>
+2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Fix build on MacOSX.
+ * 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.
-2007-06-13 Wim Taymans <wim@fluendo.com>
+2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-06-12 Wim Taymans <wim@fluendo.com>
+2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
- Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * 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.
- * 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.
+2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-11 Wim Taymans <wim@fluendo.com>
+ * 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.
- Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * ext/theora/theoraenc.c:
+ theoraenc: Fix a string leak in _getcaps()
-2007-06-07 Sebastien Moutte <sebastien@moutte.net>
+2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
-2007-06-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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/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.
+
+2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
-2007-06-07 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst/playback/gstqueue2.c: (gst_queue_get_range):
- Fix compilation.
+2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-06 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Implement GstStreamVolume interface
- Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c:
+ volume: Implement GstStreamVolume interface
-2007-06-06 Wim Taymans <wim@fluendo.com>
+2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Some more fseeko checks.
+ * 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.
-2007-06-06 Wim Taymans <wim@fluendo.com>
+2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- check for large file support.
+ * 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.
-2007-06-05 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Based on a patch by Sven Arvidsson <sa at whiz dot se>:
+ * 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
- * 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.
+2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-06-01 Michael Smith <msmith@fluendo.com>
+ * 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.
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
- Don't overflow intermediate values when seeking to large time values
- in audiotestsrc.
+2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-05 Wim Taymans <wim@fluendo.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: Add Since marker for the new skip-to-first property
- * 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.
+2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-06-05 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.
- Patch by: Edward Hervey <edward@fluendo.com>
+2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
- (gst_v4lsrc_query):
- Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
+ * 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
-2007-06-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2009-09-10 18:19:36 +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),
- (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.
+ * 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
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ playbin(2): Document that the volume property uses a linear scale
+ Fixes bug #571610.
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Back to CVS
+ * 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.
-=== release 0.10.13 ===
+2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
- * configure.ac:
- releasing 0.10.13, "What's Going on?"
+ * 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
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
- * 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/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
- Patch by: Wim Taymans <wim@fluendo.com>
- Patch by: Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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/floatcast/floatcast.h:
- Define inline when needed on win32 builds. Fixes: #441295
- Patch by: Sebastien Moutte <sebastien@moutte.net>
+2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-29 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle kate and cmml as sparse streams too
- * gst/playback/gstplaybasebin.c: (queue_overrun),
- (no_more_pads_full):
- Stop buffering when the group is commited because the queues filled up.
- Fixes #442024.
+2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-05-25 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: Better handling of sparse streams by sending segment updates
+ Fixes bug #397419.
- * 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.
+2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gsturidecodebin.c:
+ docs: tell a biit more about uri-decodebin and buffering
-2007-05-24 Wim Taymans <wim@fluendo.com>
+2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* 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.
+ 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.
-2007-05-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: add some more debug
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * ext/theora/theoraenc.c:
+ theoraenc: Print a debug message with supported formats
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
- * 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
+ * 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.
-2007-05-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * ext/theora/theoraenc.c:
+ theora: Implement support in theoraenc for Y444 and Y42B
+ Fixes bug #594165.
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
-
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * ext/theora/theoraenc.c:
+ theora: Refactor the buffer copy code
- * 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.
+2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theora: Split yuv_buffer creation into its own function
- * ext/cdparanoia/gstcdparanoiasrc.c:
- Repair umlaut.
+2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
-2007-05-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/theoraenc.c:
+ theora: Split out buffer resize in its own function
- * 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.
+2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
-2007-05-21 Jan Schmidt <thaytan@mad.scientist.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.
- * 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.
+2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-21 Michael Smith <msmith@fluendo.com>
+ * tests/examples/seek/seek.c:
+ seek: make stop state configurable
+ Make it easy to experiment with different stop states (NULL and READY)
- * ext/theora/theoradec.c: (theora_handle_data_packet):
- Don't look inside 0-length packets (which indicate duplicated
- frames)
+2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-21 Wim Taymans <wim@fluendo.com>
+ * 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
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_read_sector):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Small cleanups.
+2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix typo.
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ baseaudiosink: whitespace fixes
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- Add some FIXME
+2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (queue_underrun_cb):
- And some debug info when a FIXME path is hit.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add more debug
-2007-05-21 Wim Taymans <wim@fluendo.com>
+2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@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_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/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ whitespace fixes
- * 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.
+2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_change_state):
- Fix typo in comment.
+2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
- * gst/volume/Makefile.am:
- Fix tabs.
+2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+ * 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()
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
- set_option, get_option, _gst_reserved):
- Revert reordering functions (keep ABI).
+ * 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.
-2007-05-17 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
- * 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.
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix compiler warning with MSVC
+ Fixes bug #594275.
- Fixes: #393975
+2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
- Clean up the X event handling loop and make them the same for
- both xvimagesink and ximagesink while I'm at it.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: fix event forwarding
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
- Make decodebin2 autoplug depayloaders too.
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
+ Fixes #594258
- * gst/playback/gsturidecodebin.c: (source_new_pad):
- Set the newly created decoder in a usable state when autoplugging a
- dynamic source such as RTSP.
+2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-17 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ fix whitespace
- * 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.
+2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
-2007-05-17 Wim Taymans <wim@fluendo.com>
+ * 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/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.
+2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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>
+ * 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.
- * 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.
+2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
-2007-05-17 Wim Taymans <wim@fluendo.com>
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix small memory leak when setting window title
- * 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.
+2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
-2007-05-17 Wim Taymans <wim@fluendo.com>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix small memory leak when setting window title
- * 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.
+2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 David Schleef <ds@schleef.org>
+ * .gitignore:
+ introspection: Add *.gir and *.typelib to .gitignore
- * docs/Makefile.am: Install docs even when --disable-gtk-doc
- is disabled. This matches the behavior of gtk+. Fixes #349099.
+2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 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
- * 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.
+2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Fix introspection build by ordering sources/headers in dependency order
- * 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.
+2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Remove debug echo
- * configure.ac:
- We don't require a C++ compiler. So don't require one.
+2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix build of introspection data by using dependency order for the headers/sources
- * 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.
+2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-15 Stefan Kost <ensonic@users.sf.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: Strip Gst prefix from all types/functions
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * 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
-2007-05-14 David Schleef <ds@schleef.org>
+2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add support for video/x-raw-bayer.
+ * gst-libs/gst/video/Makefile.am:
+ video: Add gobject-introspection support
-2007-05-12 David Schleef <ds@schleef.org>
+2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Add some sanity checking for the XVImage size returned by X.
- Related to #377400.
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Add gobject-introspection support
-2007-05-12 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: Add gobject-introspection support
-2007-05-12 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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
-2007-05-11 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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):
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Add gobject-introspection support
- 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.
+2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Also cleanup the MPEG elementary and MPEG-TS detection functions a
- little.
+ * gst-libs/gst/rtp/Makefile.am:
+ rtp: Add gobject-introspection support
- Tested against my media test directory, with some improvements and
- no regressions.
+2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-10 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/riff/Makefile.am:
+ riff: Add gobject-introspection support
- * 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.
+2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-09 Sebastien Moutte <sebastien@moutte.net>
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: Add gobject-introspection support
- * 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.
+2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-09 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/netbuffer/Makefile.am:
+ netbuffer: Add gobject-introspection support
- * ext/theora/theoradec.c: (theora_handle_type_packet):
- gst_pad_push_event() does not return a GstFlowReturn!
+2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-09 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/interfaces/Makefile.am:
+ interfaces: Add gobject-introspection support
- * tests/examples/seek/scrubby.c: (stop_cb), (main):
- * tests/examples/seek/seek.c: (do_seek):
- Some small cosmetic changes.
+2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-08 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/fft/Makefile.am:
+ fft: Add gobject-introspection support
- * 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.
+2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-07 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/cdda/Makefile.am:
+ cdda: Add gobject-introspection support
+ This is disabled for now until gobject-introspection is fixed
- * 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.
+2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Add gobject-introspection support
- * 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.
+2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * gst-libs/gst/app/Makefile.am:
+ app: Add gobject-introspection support
- * 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.
+2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 00a859e to 19fa4f3
- * 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.
+2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track):
- Fix build when disabling asserts.
+2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-03 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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).
+2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-03 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-03 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: add some debugging
- * 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.
+2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-03 Edward Hervey <edward@fluendo.com>
+ * 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.
- * 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.
+2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-05-01 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/playback/gstdecodebin.c: (type_found), (plugin_init):
- * gst/playback/gstdecodebin2.c: (plugin_init):
- Better error message for text files.
+2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppay: add some debugging
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
- Fix offset bug in generation RR packets.
+2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-27 Julien MOUTTE <julien@moutte.net>
+ * 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>
- * 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.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: move function around
-2007-04-27 Wim Taymans <wim@fluendo.com>
+2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix sample duration calculation
-2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@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/rtp/gstbasertpaudiopayload.c:
+ audiortppay: more refactoring
+ Unify the sample/frame buffer handling code by making the functions plugable.
-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.
+2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-24 Tim-Philipp Müller <tim at centricular dot 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.
- Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
+2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c:
- Fix build if XShm is not available (#432362).
+ * 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()
-2007-04-24 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@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.
+ * common:
+ Update common
-2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Dan Williams <dcbw redhat com>
+ * 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/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.
+2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-23 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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
+2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppay: add macro to cast
- Patch by: Olivier Crete <tester at tester ca>
+2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@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).
+ audiopayload: code cleanups
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@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.
+ audiortppayload: don't check adapter
+ the adapter is never NULL so we don't need to check it.
+ Use _scale functions to avoid overflows.
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Zeeshan Ali <zeenix gmail com>
+ * 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/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).
+2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/subparse/gstsubparse.c:
+ subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
- * gst/audioresample/gstaudioresample.c:
- Make more functions static, just because we can.
+2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-21 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.
- * tests/check/elements/audioresample.c:
- Add unit test for audioresample shutdown crasher (#420106).
+2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-20 Stefan Kost <ensonic@users.sf.net>
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: don't log not-yet-initialised integer value
- * gst/subparse/gstsubparse.c:
- * gst/subparse/samiparse.c:
- Use GST_DISABLE_XML here
+2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@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
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
+ And reflow code to be more indent friendly.
-2007-04-19 Michael Smith <msmith@fluendo.com>
+2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
- * 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-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.
-2007-04-18 Stefan Kost <ensonic@users.sf.net>
+2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/pipelines/.cvsignore:
- Fix build.
+ * 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.
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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 Sebastian Dröge <slomo@circular-chaos.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.
-
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.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.
-
- 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.
-
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
-
- 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.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post missing plugin messages before any error messages
-2007-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: safely handle the indexes
-2007-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * win32/common/libgstrtsp.def:
+ def: add new rtsp symbols
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: whitespace fixes.
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
- * 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/gdp/gstgdppay.c:
+ Bug 593035 - set IN_CAPS for streamheader buffer
-2007-04-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix misleading comment.
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: The internally linked pad of the selector might be NULL in some cases
-2007-04-13 Stefan Kost <ensonic@users.sf.net>
+2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- More sanity checks for the header fields.
+ * 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.
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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-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.
-2007-04-12 Wim Taymans <wim@fluendo.com>
+2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
- (gst_video_rate_chain):
- Add some debug.
+ * 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.
- * tests/check/elements/videorate.c: (GST_START_TEST),
- (videorate_suite):
- Added check for videorate changing caps handling. Closes #421834.
+2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-12 Michael Smith <msmith@fluendo.com>
+ * tests/check/elements/audioresample.c:
+ audioresample: Add unit test for checking for timestamp drifts
+ This also checks for perfect timestamping and offsetting.
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Use scale functions to avoid overflow when calculating duration of
- vorbis buffers.
+2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
- Use gst_tag_freeform_string_to_utf8() here.
+ * tests/check/elements/audioresample.c:
+ audioresample: Improve debugging a bit in the unit test
-2007-04-12 Wim Taymans <wim@fluendo.com>
+2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
- (gst_gdp_pay_sink_event):
- Make sure we set the IN_CAPS flag correctly.
+ * 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.
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Get the IN_CAPS flag before we call functions that mess with the flags.
+2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
-2007-04-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * gst/subparse/gstsubparse.c:
+ subparse: Remove dependency on regex.h as it's not used anyway
+ Fixes bug #592544.
- * 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
+2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-04-06 Wim Taymans <wim@fluendo.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer overflow when pushing the drain
- * gst/playback/gstplaybin.c: (add_sink),
- (gst_play_bin_change_state):
- Activate sync in playbin, we are ready to handle it for live streams.
+2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-04-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: Fix timestamp drift
+ Fixes bug #591934.
- * tests/check/elements/playbin.c:
- (test_sink_usage_video_only_stream), (playbin_suite):
- Add small test for stream-info-value-array code paths.
+2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
-2007-04-05 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-05 Wim Taymans <wim@fluendo.com>
+ * 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
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.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.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Mark Transport as supporting multiple values.
-2007-04-03 David Schleef <ds@schleef.org>
+2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added missing Since tags.
-2007-04-03 Wim Taymans <wim@fluendo.com>
+2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * 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
- * 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.
+2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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>
- * 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-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made gst_rtsp_watch_queue_data() thread safe.
-2007-03-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.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-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()
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
- Patch by: René Stadler <mail at renestadler dot de>
- with some minor changes
+ * 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-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.
+2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * 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).
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * 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.
-2007-03-29 Andy Wingo <wingo@pobox.com>
+2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
- perfect offsets also, not just timestamps.
+ * 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.
- * tests/check/elements/videorate.c (test_more): Test that given
- any incoming offsets, that videorate produces perfect offsets.
+2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improve parse_line().
+ Make parse_line() handle keys with multiple values on one line correctly.
- * gst-libs/gst/riff/riff-ids.h:
- Add some more RIFF formats.
+2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-27 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * 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.
+2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-27 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.
- * 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.
+2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-26 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Normalize lines (remove extra whitespace) before parsing.
- Patch by: René Stadler <mail at renestadler de>
+2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made parse_string() return a result.
+ This will catch parsing errors when a too long string is received.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit tests for replaygain tags in vorbis comments (closes #423055).
+2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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/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.
+2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * gst-libs/gst/rtp/Makefile.am:
- The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
+2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-23 Michael Smith <msmith@fluendo.com>
+ * 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()
- * 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.
+2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * 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.
+2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-22 Wim Taymans <wim@fluendo.com>
+ * 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
- * 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.
+2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: fix includes for appsrc/appsink
- * gst/playback/gstplaybin.c: (add_sink):
- Use GST_PTR_FORMAT to log caps.
+2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+ * 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).
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@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/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.
-2007-03-20 Michael Smith <msmith@fluendo.com>
+2009-08-18 21:45:31 +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.
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
-2007-03-19 Wim Taymans <wim@fluendo.com>
+2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/playback/gstdecodebin.c: (remove_fakesink):
- Don't post STATE_DIRTY anymore.
+ * 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).
- * 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.
+2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-17 David Schleef <ds@schleef.org>
+ * 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...
- * REQUIREMENTS: Update this file, change the formatting to make
- it more consistent, plus more machine readable.
+2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-16 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playsink: Also send SEEK events directly to a subpicture sink
- * 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.
+2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-16 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-15 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * 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.
- Patch by: Olivier Crete <olivier.crete@collabora.co.uk>
+2009-08-13 15:48:00 +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/playback/gsturidecodebin.c:
+ uridecodebin: Post a missing plugin message additional to the error message on unknown types
+ Fixes bug #591677.
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * 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:
- Update translations from translation project.
+ * po/zh_CN.po:
+ playbin2: fix error message string
+ Fixes #591577.
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@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-libs/gst/riff/riff-read.c:
+ riff: align API doc of gst_riff_parse_chunk with reality
-2007-03-13 Michael Smith <msmith@fluendo.com>
+2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@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...
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid assertion failure on empty/NULL caps
-2007-03-10 Sebastien Moutte <sebastien@moutte.net>
+2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Also detect SVG by the <svg> starting tag
+ Not all SVG images have the DOCTYPE specified.
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@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/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.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit test for the above.
+2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/pipelines/streamheader.c:
+ streamheader: Fix caps leak in the vorbisenc unit test
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add unit test for MPL2 subtitle format (#413799).
+2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-10 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.
- Patch by: Kamil Pawlowski <kamilpe gmail com>
+2009-08-10 10:17:07 +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-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
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
- * configure.ac:
- We require core CVS for the new buffer metadata copy functions.
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix doc blurb typo for gst_navigation_send_key_event
-2007-03-09 Wim Taymans <wim@fluendo.com>
+2009-08-09 12:13:16 +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/subparse/gstsubparse.c:
+ subparse: Allow . instead of , as millisecond delimiter in srt subtitles
+ Fixes bug #591207.
- Patch by: Alex Lancaster <alexl at users sourceforge net>
+2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * 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
- * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
- (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
- Improve adapter usage and comments.
+2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
- Basetransform copied the metadata for us.
+ * 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.
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/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
- * 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.
+2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
-2007-03-09 Michael Smith <msmith@fluendo.com>
+ * docs/design/draft-va.txt:
+ docs: add draft for generic introduction of video acceleration APIs idea
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
- Replace AC3 typefinder with one that isn't terrible, and actually
- works usefully.
+2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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).
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- fix error category and translatable string
-
+2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Fix up utils => pbutils here too.
+2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-09 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: Convert theoradec to libtheora 1.0 API
- * 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)
+2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextrender.c:
+ textrender: Fix blitting of text over the output buffer and cairo painting
- * tests/check/Makefile.am:
- Don't test libvisual elements in the states check, because libvisual
- seems to leak internally.
+2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Re-enable the alsa and states tests now that there's new suppressions
- in gst.supp.
+ * ext/pango/gsttextrender.c:
+ textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
- * tests/check/elements/alsa.c: (GST_START_TEST):
- Don't leak the alsamixer we instantiated.
+2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/icles/test-colorkey.c:
+ colorkey-test: fix xsync error
- * 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.
+2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Extra log line.
+2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
- * 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.
+2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/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).
+2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: change default slave method
+ Set the default slave method to the much better skew slaving algortihm.
- * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
- Unit test to check the above.
+2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-07 Wim Taymans <wim@fluendo.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: make buffer writable
+ Make the input buffer writable before changing its contents.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
- Fix confusing debug message.
+2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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-plugins-base.doap:
- update doap file with new version
+2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextoverlay.c:
+ pango: Send queries from the srcpad directly to the video sinkpad
- * configure.ac:
- Back to CVS
+2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.12 ===
+ * gst/subparse/gstsubparse.c:
+ subparse: Implement POSITION query
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- releasing 0.10.12, "Zombie Horde"
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ subparse: Implement SEEKING query
-2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
* configure.ac:
- Bump version to 0.10.11.4 pre-release
+ * 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.
-2007-03-06 Wim Taymans <wim@fluendo.com>
+2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ pango: Add support for xRGB and BGRx formats
-2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Bump version to 0.10.11.3 pre-release
+ * 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.
-2007-03-05 Sebastian Dröge <slomo@circular-chaos.org>
+2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/POTFILES.in:
- Update paths for the rename from utils to pbutils to fix the build.
+ * ext/pango/gsttextoverlay.c:
+ pango: Re-add shading support which was dropped by a previous patch
-2007-03-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/pbutils/Makefile.am:
- Change directory to install headers in from gst/utils to gst/pbutils
- as well.
+ * configure.ac:
+ * ext/pango/gsttextoverlay.c:
+ pango: Check if pangocairo supports vertical rendering and fix properties
-2007-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/pango/gsttextrender.c:
+ textrender: Use PROP_X instead of ARG_X consistently
-2007-03-02 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 11:42:28 +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/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.
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ pango: Some minor cleanup
-2007-03-01 Jan Schmidt <thaytan@mad.scientist.com>
+2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
- Update to 0.10.11.2 (0.10.12 pre-release)
+ pango: Check for pangocairo instead of pangoft2
-2007-03-01 Wim Taymans <wim@fluendo.com>
+2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- base time is irrelevant here.
+ * 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.
-2007-03-01 Wim Taymans <wim@fluendo.com>
+2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
- * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
- Improve debugging.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add SVG typefinder
- * 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.
+2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full):
- When we are asked to render N sample to 0 bytes, return N.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add postscript typefinder
-2007-03-01 Wim Taymans <wim@fluendo.com>
+2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Use static caps again for MPEG4 typefinding
- * 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.
+2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Implement better & more flexible MPEG4 typefinding
+ This detects more MPEG4 streams as MPEG4.
+ Fixes bug #556537.
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Convert to new AG_GST style.
+ * 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.
-2007-02-28 Wim Taymans <wim@fluendo.com>
+2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Ed Catmur <ed at catmur dot co dot uk>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Restrict width/height to 2^15 - 1
+ Otherwise integer overflows will happen, resulting in segmentation faults.
+ Fixes bug #590243.
- * 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.
+2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-28 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix indention of template header
- * tests/check/Makefile.am:
- Include local stuff before system installed things in LDFLAGS and
- CFLAGS.
+2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clarify documentation about caps and linkage
+ Fixes bug #589095.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
- Improve debugging.
+2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix typefinding of SDP files
+ Fixes bug #589574.
- * 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.
+2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Take the output offsets from the input if possible
+ Fixes bug #588915.
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
- (gst_audio_clock_new):
- Fix clock name.
+2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Make sure to allocate enough memory for the temporary buffer
+ and fix scaling of odd-height interlaced video.
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@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
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix interlaced scaling for I420
+ ...and some other minor mistakes in the previous change.
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@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.
+ * 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.
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@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.
+ * 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.
-2007-02-27 Julien MOUTTE <julien@moutte.net>
+2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.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).
+ * configure.ac:
+ back to development -> 0.10.24.1
-2007-02-26 Wim Taymans <wim@fluendo.com>
+2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/examples/seek/seek.c: (stop_seek):
- When we stop scrubbing, don't leave the pipeline PLAYING when we
- requested a PAUSED state.
+ * gst-plugins-base.doap:
+ Add 0.10.24 release to the doap file
-2007-02-25 Tim-Philipp Müller <tim at centricular dot net>
+=== release 0.10.24 ===
- Patch by: René Stadler <mail at renestadler de>
+2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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).
+ * 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
- * tests/check/libs/tag.c: (GST_START_TEST):
- Test case for the above.
+2009-08-05 00:38:40 +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/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
-2007-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Loïc Minier <lool+gnome at via ecp fr>
+ * 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.
+
+2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
* configure.ac:
- * ext/alsa/Makefile.am:
- * gst/audiotestsrc/Makefile.am:
- Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
+ * 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
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Improve docs: point out that the application needs to assist playbin
- with buffering.
+ * 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.
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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).
+ * gst/adder/gstadder.c:
+ adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
-2007-02-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
- * tests/check/Makefile.am:
- Enable unit test for GstNetBuffer.
+2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
-2007-02-22 Andy Wingo <wingo@pobox.com>
+ * 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.
- * 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.
+2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-22 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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
+2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * 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.
+2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/examples/v4l/.gitignore:
+ ignores: Ignore v4l probing example binary
- * tests/examples/seek/seek.c: (do_seek):
- Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
+2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * gst/volume/gstvolume.c: (volume_process_int16),
- (volume_process_int16_clamp), (volume_set_caps):
- Unbreak volume, value remains gint.
+2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+ * common:
+ Automatic update of common submodule
+ From fedaaee to 94f95e3
- * 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.
+2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
-2007-02-20 Edward Hervey <edward@fluendo.com>
+ * gst-plugins-base.spec.in:
+ Update spec file with latest changes
- * 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.
+2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-02-20 Edward Hervey <edward@fluendo.com>
+ * 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
- * 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
+2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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).
+2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiosink: Add stream-status messages
+ Fixes #587695
- * gst-libs/gst/audio/audio.c:
- Fix documentation.
+2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: Add stream-status messages
+ See #587695
- Patch by: Yves Lefebvre <ivanohe abacom com>
+2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
- Don't leak caps. Fixes #408278.
+ * gst/adder/gstadder.c:
+ gstadder: Don't forget to free pending events on flush/dispose.
+ Fixes #588747
-2007-02-15 Stefan Kost <ensonic@users.sf.net>
+2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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)
+ * tests/check/elements/adder.c:
+ tests/adder: Add stream consistency checking. Fixes #588748
-2007-02-15 Wim Taymans <wim@fluendo.com>
+2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
- * sys/ximage/ximagesink.c:
- (gst_ximagesink_calculate_pixel_aspect_ratio):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_calculate_pixel_aspect_ratio):
- Small constifications.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Make sure tags are properly serialized. Fixes #588746
+ We do this by letting the basesrc base class handle the tags.
-2007-02-15 Wim Taymans <wim@fluendo.com>
+2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: Collect incoming tag events and send them after newsegment. Fixes #588747
- * 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.
+2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-02-15 Wim Taymans <wim@fluendo.com>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Check for empty tag strings. Fixes #588724
- * 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.
+2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-13 Stefan Kost <ensonic@users.sf.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
- * configure.ac:
- Remove 'tests/examples/xerror/Makefile' from output files again.
+2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-13 Stefan Kost <ensonic@users.sf.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.
- * configure.ac:
- * docs/plugins/Makefile.am:
- Also crossref against gst-plugins-base-libs.
+2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-12 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * configure.ac:
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Add crossreferences to glib/gobject/gstream docs.
+2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * gst-libs/gst/audio/audio.h:
- Source formatting.
+ * tests/examples/v4l/Makefile.am:
+ examples: fix v4l probe example build
+ Fixes bug #588550.
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
- Add own debug category.
+2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-02-12 Tim-Philipp Müller <tim at centricular dot 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/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.23.2 pre-release
- Patch by: René Stadler <mail at renestadler de>
+2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/tag/gstvorbistag.c:
- Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
- (#403597).
+ * po/LINGUAS:
+ * po/tr.po:
+ Add Turkish translations
-2007-02-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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>
+ * 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.
- * 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.
+2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-02-10 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.
- * 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).
+2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/check/elements/subparse.c:
- Add test for this.
+ * 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.
-2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
- * 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).
+2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2007-02-09 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/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).
+2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-02-08 Tim-Philipp Müller <tim at centricular dot net>
+ * 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://...
- Based on patch by: Julien Puydt <julien.puydt at laposte net>
+2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.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.
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: simplify docs
+ Make short-desc short - its used in the toc. Strip uneeded markup.
-2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/audioconvert/gstaudioconvert.c:
- Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
+ * 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
-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>
+2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/audioconvert/gstaudioconvert.c:
- Also mention that a conversion from double to float is suboptimal still.
+ * tests/examples/gio/.gitignore:
+ ignores: ignore the giosrc-mounting example binary
-2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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/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.
-2007-02-05 Andy Wingo <wingo@pobox.com>
+2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@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-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Siren codec and make two descriptions non-translatable
-2007-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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).
+ * common:
+ Automatic update of common submodule
+ From 5845b63 to fedaaee
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
- * 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.
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add siren to the RIFF parser
+ Add siren7 caps to the RIFF parser.
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * 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.
+ * 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
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * 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:
+ * 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.
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst/adder/gstadder.c:
+ adder: add since tags to docs
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-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.
+2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@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).
+ * tests/examples/seek/seek.c:
+ seek: add a checkbox for progressive download
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Build and document new API.
+2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@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/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.
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@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/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.
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-2007-02-02 Stefan Kost <ensonic@users.sf.net>
+2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@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/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.
-2007-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
- Patch by: Holger Wansing <linux wansing-online de>
+ * 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.
- * po/LINGUAS:
- * po/de.po:
- Add German translation (#352069).
+2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
-2007-02-01 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- reviewed by: Wim Taymans <wim@fluendo.com>
+2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@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/gio/gstgiosink.c:
+ gio: Also post a "not-mounted" message from giosink
-2007-01-31 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/examples/gio/giosrc-mounting.c:
+ gio: Remove workaround for playbin2 bug in the sample application
+ The playbin2 bug was #588078.
- Fixes: #402505
+2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-30 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-30 Wim Taymans <wim@fluendo.com>
+ * 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
- * gst/audioconvert/gstaudioconvert.c:
- Don't fail on 0 sized buffers. Fixes #396835.
+2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 David Schleef <ds@schleef.org>
+ * configure.ac:
+ gio: Remove the experimental status from the GIO plugin
+ Fixes bug #510417.
- * gst/typefind/gsttypefindfunctions.c:
- Detect BBCD as video/x-dirac, so we can play raw dirac
- streams.
+2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ gio: Add documentation for the new "not-mounted" and "file-exists" messages
- * 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.
+2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgiobasesrc.c:
+ gio: Make sure that we have the correct stream position when starting
- * tests/check/gst/.cvsignore:
- Do as buildbot says.
+2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 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.
- * 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.
+2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * 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.
-2007-01-27 Wim Taymans <wim@fluendo.com>
+2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/check/pipelines/gio.c:
+ gio: Try to reuse the pipeline with the same stream objects
-2007-01-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesrc.c:
+ gio: Improve the error message if a stream is already closed before usage
-2007-01-25 Wim Taymans <wim@fluendo.com>
+2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (remove_source):
- Don't try to disconnect a signal from a finalized object.
+ * 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.
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/gio/gstgiosrc.c:
+ gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
-2007-01-25 Edward Hervey <edward@fluendo.com>
+2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * ext/gio/gstgiosink.c:
+ gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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().
+ * 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.
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Make getting the current frame thread-safe.
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: Fix compilation when debugging is disabled
+ Fixes bug #587980.
-2007-01-25 Edward Hervey <edward@fluendo.com>
+2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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
-2007-01-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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
- * tests/check/gst-plugins-base.supp:
- Add suppression for pango on edgy/x86 for textoverlay test.
+2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-24 Wim Taymans <wim@fluendo.com>
+ * 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
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Add some more fixed payloads.
+2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-23 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
-2007-01-23 Tim-Philipp Müller <tim at centricular dot 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.
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
- Don't leak mutex.
+2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * tests/check/elements/adder.c:
+ adder: make test more robust
+ Add audioconverts to the live-seeking test to make it negotiate.
-2007-01-23 Edward Hervey <edward@fluendo.com>
+2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: use core performance log category
-2007-01-23 Julien MOUTTE <julien@moutte.net>
+2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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.
-2007-01-22 Edward Hervey <edward@fluendo.com>
+2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
- * 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.
+ * 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.
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
- * 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>.
+ * ext/Makefile.am:
+ build: remove spurious schroedinger reference
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Lock that X Call as well. Fixes #397673.
+ * 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>
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ netaddress: add constant for max len
- * 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.
+2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/gst/.cvsignore:
- * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
- (typefindfunctions_suite):
- Add unit test for the above.
+ * 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()
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- On second thought, use "depth" field rather than "bpp" field.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: make fd:// uri use buffering too
+ fd:// usually operate in push mode only and are thus suitable for buffering.
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Camtasia caps apparently need a bpp field (#398875).
+ * gst/playback/gstplaybin2.c:
+ * gst/volume/gstvolume.c:
+ volume: include "1.0=100%" in property description
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/playback/gstplaysink.c:
+ playsink: remove unused property defs
- * 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).
+2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/multichannel.c:
+ multichannel: rewrite the new doc comment a bit
+ Its part of the audio lib.
- * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
- Plug a few reference leaks.
+2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot 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).
- * 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.
+2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-18 Tim-Philipp Müller <tim at centricular dot 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'
- * gst/audioconvert/gstchannelmix.c:
- Remove compatibility cruft for prehistoric GLib versions.
+2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-17 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: include in.h for FreeBSD compat
+ Fixes #586920
- * 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).
+2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-16 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstapp.def:
+ defs: add defs for new appsink buffer-list method
- * 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).
+2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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.
+2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/appsink.c:
+ Added unit tests for buffer list support in appsink.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Red and blue mask was swapped (spotted by Dan Williams).
+2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2007-01-15 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsink.c:
+ Added buffer list support.
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Use new beats-per-minute tag from core.
+2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2007-01-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsink.h:
+ Added buffer list support.
- * po/POTFILES.in:
- Add new files with translatable strings, so they actually make it
- into the template file one day.
+2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-01-12 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Include winsock2.h after defining WINVER.
+ Similar to bug #587080.
- * 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.
+2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
- of a pad function.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Moved a comment.
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
- * gst/typefind/gsttypefindfunctions.c:
- Fix comment.
+ * 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.
-2007-01-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
- * 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).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Define WINVER before including any win headers
+ Fixes bug #587080.
-2007-01-10 Wim Taymans <wim@fluendo.com>
+2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
- * 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.
+ * 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.
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@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/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.
- * 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.
+2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-09 Tim-Philipp Müller <tim at centricular dot 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
- * 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.
+2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Add new lib.
+ * common:
+ Automatic update of common submodule
+ From f810030 to 5845b63
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Generate docs for new lib and API.
+2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst/playback/gstscreenshot.c:
+ screenshot: don't leak message
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/Makefile.am:
- Dist gstoggdemux.h to fix 'make distcheck'.
+ * gst/typefind/gsttypefindfunctions.c:
+ 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.
- * sys/v4l/Makefile.am:
- Fix 'make distcheck' even more.
+2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
-2007-01-09 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From f3bb51b to f810030
- * 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.
+2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-09 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for multipart
+ So we get slightly nicer error messages when multipartdemux is missing.
- * 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
+2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * configure.ac:
- Require core CVS HEAD for Andy's basesrc/sink API additions.
+2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix leak when the source fails to change state
- Patch by: Günter Thelen <daedalus dot inc at gmx net>
+2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * gst/subparse/gstssaparse.c:
+ ssaparse: avoid leaking all buffers
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Check if localtime_r() is available.
+ * 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)
- * 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).
+2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-01-08 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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:
+ * 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.
-2007-01-07 Julien MOUTTE <julien@moutte.net>
+2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
+ Also make all the function arrays constant.
- * 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:
+2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-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/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.
-2007-01-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
- (gst_ogm_parse_change_state):
- Activate pads before adding them to the element.
+ * 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.
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/examples/seek/scrubby.c: (main):
- * tests/examples/seek/seek.c: (main):
- Call g_thread_init() first thing in main() (see #391278).
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix NV12 painting for odd strides/heights
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Update to use GST_PLUGINS_BASE_CFLAGS as well.
+ * 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.
-2007-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
* 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
+ * 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.
-2007-01-04 Julien MOUTTE <julien@moutte.net>
+2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
- * 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 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-plugins-base.spec.in:
+ Make build of schro plugin conditional
- * 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).
+2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-01 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * configure.ac:
- * docs/Makefile.am:
- * docs/design/Makefile.am:
- Dist design docs.
+2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+ * 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...
- * docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation
- typo. Fixes: #390063.
+2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+ * win32/common/libgstrtp.def:
+ defs: add new symbol to win32 defs file
+ Based on patches by Ognyan Tonchev.
+ See #585559
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
- caps leak.
- * win32/common/config.h: Updated.
+2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-22 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * 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
+2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * win32/common/libgstrtp.def:
+ rtp: cleanups
+ Add Since tags to docs
+ Move some code around
+ Add win32 symbols
- * 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.
+2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Add some inconsequential branch hints in a couple of places.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtp: add bufferlist support
-2006-12-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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)
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: pass data to macros instead of GstBuffer
-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
+2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-12-20 Tim-Philipp Müller <tim at centricular dot 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
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Forgot to register the extensions.
+2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-12-20 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: appname might be NULL
+ Don't set title if appname is unknown.
- * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
- (plugin_init):
- Add typefinder for VIVO files (my christmas present to the 90s).
+2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: set window title from application name
- * 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.
+2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
- 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.
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Made the parsing of the RTSP URL scheme more generic.
- 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).
+2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
- Same problem also needs to be fixed for playbin+decodebin2.
+ * 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()
- * 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.
+2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Only extract the session ID from RTSP responses.
- * 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.
+2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-12-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Small docs fixes/updates.
+2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
-2006-12-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
- * Makefile.am:
- * gst-plugins-base.doap:
- * gst-plugins-base.spec.in:
- add doap file
+ * 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.
-2006-12-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Jens Granseuer <jensgr at gmx net>
+ * 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
- * 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.
+2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * configure.ac:
- Bump version nano - back to CVS.
+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
-=== release 0.10.11 ===
+2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
-2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: add some extra logging for buffer timestamps
- * configure.ac:
- releasing 0.10.11, "Dumb things"
+2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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/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/playback/gstdecodebin2.c:
+ decodebin2: Free iterator after removing all groups
-2006-12-04 Wim Taymans <wim@fluendo.com>
+2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix #382223, add more dynamic caps handling.
+ * 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.
-2006-12-04 Wim Taymans <wim@fluendo.com>
+2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@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-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add Timestamp header field
+ fixes #585994
-2006-12-01 Michael Smith <msmith@fluendo.com>
+2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Delete bad debug code.
- Fixes #381219
+ * 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
-2006-12-01 Jan Schmidt <thaytan@mad.scientist.com>
+2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * gst/playback/gstplaysink.c:
+ playsink: set the sink flag on the element
-2006-11-30 Michael Smith <msmith@fluendo.com>
+2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/playback/gsturidecodebin.c:
+ uridecodebin: add debug message
-2006-11-28 Wim Taymans <wim@fluendo.com>
+2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (check_queue_event):
- Improve debug.
+ * 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.
- * 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.
+2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-11-28 Wim Taymans <wim@fluendo.com>
+ * 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).
- * 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.
+2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-11-28 Edward Hervey <edward@fluendo.com>
+ * 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.
- * 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.
+2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-11-28 Wim Taymans <wim@fluendo.com>
+ * tests/examples/seek/seek.c:
+ Don't use deprecated GTK API
+ Fixes bug #585758.
- * 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.
+2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-26 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use a more consistent naming of GstRTSPRec variables.
-2006-11-24 Wim Taymans <wim@fluendo.com>
+2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * 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.
-2006-11-24 Wim Taymans <wim@fluendo.com>
+2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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:
+ * 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.
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.h:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- remove obsolete _factory_init protos
+ * 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.
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
- Fix spacing in debug message.
+ * 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)
-2006-11-23 Wim Taymans <wim@fluendo.com>
+2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * tests/examples/seek/seek.c:
+ seek: add shuttle controls
-2006-11-23 Michael Smith <msmith@fluendo.com>
+2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@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.
+ * tests/examples/seek/stepping2.c:
+ example: fix compile
-2006-11-22 Michael Smith <msmith@fluendo.com>
+2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/examples/seek/Makefile.am:
+ examples: build the stepping2 example
- * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
- (GST_START_TEST), (theoraenc_suite):
- Enable discontinuity test, fix it.
+2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playsink: update for new step API
- * 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.
+2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-11-21 Stefan Kost <ensonic@users.sf.net>
+2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Add some new symbols to the docs
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: handle border cases in resampler
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.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).
+ * common:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ docs: Update common. Use upload-doc.mak instead of upload.mak
-2006-11-20 Wim Taymans <wim@fluendo.com>
+2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix RGBA32 caps. Fixes #357038.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ docs: fix typo
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/mixertrack.h:
- Add FIXME so we can add some padding here in 0.11
+ * 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.
-2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.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).
+ * 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.
-2006-11-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/testchannels.c:
+ Don't include config.h multiple times when build audio testchannel app.
+ Fixes build problem on win32 (#585075).
- * 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.
+2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-11-16 Michael Smith <msmith@fluendo.com>
+ * 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)
- * 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.
+2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/Makefile.am:
- Enable audiorate test now that it passes.
+ * 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.
-2006-11-09 Stefan Kost <ensonic@users.sf.net>
+2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- clear xv when going to NULL, remove // commented non-existant proto
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: add some more docs
- * tests/examples/seek/seek.c: (main):
- add missing tooltip description for scrub and play_scrub
+2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-11-14 David Schleef <ds@schleef.org>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: Avoid a compiler warning.
- * 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.
+2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-11-14 David Schleef <ds@schleef.org>
+ * 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.
- * 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.
+2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-11-14 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ autogen: remove -Wno-portability from here
+ as it is in configure.ac now.
- Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
+2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Plug a memory leak.
+ Free memory related to any partially read and/or written RTSP messages.
-2006-11-13 Wim Taymans <wim@fluendo.com>
+2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* 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.
+ 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.
-2006-11-13 Michael Smith <msmith@fluendo.com>
+2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
- Improve a debug line slightly.
+ * 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.
- * 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).
+2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-11-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Sergey Scobich <sergery.scobich at gmail com>
+2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
- * win32/common/libgsttag.def:
- Add missing symbol (#366492).
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for hdv/aux-* formats.
-2006-11-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
- * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
- Don't unref a NULL pad.
+ * ext/schroedinger/Makefile.am:
+ Added libgstbase to schro's LIBADD
+ Fixes #585079
-2006-11-09 Wim Taymans <wim@fluendo.com>
+2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-11-07 Stefan Kost <ensonic@users.sf.net>
+2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: make sure varargs are of right type
+ Explicitly cast the variables to g_object_set to their right types.
-2006-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-11-03 David Schleef <ds@schleef.org>
+2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- Add/remove KW-DIRAC header here, since it is ogg-specific.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Fixed a typo.
-2006-11-03 Michael Smith <msmith@fluendo.com>
+2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
- Recognise more mpeg4 elementary video streams.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Remove an unused variable.
-2006-11-02 Edward Hervey <edward@fluendo.com>
+2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Removed duplicate initialization of conn->writefd.
-2006-11-02 Wim Taymans <wim@fluendo.com>
+2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use #defined status codes.
+
+2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-11-02 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Sergey Scobich <sergey.scobich at gmail com>
+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/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).
+ * 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.
-2006-11-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstogg.c:
- Remove unused variable.
+ * gst/playback/gstplaybin2.c:
+ playbin2: API: Add {audio,video,text}-tags-changed signals
+ Fixes bug #584686.
- * ext/ogg/gstoggdemux.c:
- Fix Wim's surname in plugin description.
+2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-31 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst-plugins-base.spec.in:
- spec new .h file. Fixes #368310.
+2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-31 Michael Smith <msmith@fluendo.com>
+ * tests/examples/seek/.gitignore:
+ ignore: add new stepping app to ignore
- * 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.
+2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-31 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_init):
- Fix and activate base audio payloader.
+2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
- (plugin_init):
- Add typefinder for QuickTime Image Files (see #366156).
+2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: add G_LIKELY because we can
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Another typo fix (#366212).
+2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-10-27 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix caps for ogg typefinder.
- * 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.
+2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-27 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: remove some cruft from -sections.txt file
- Patch by: Mark Nauwelaerts <manauw at skynet dot be>
+2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
- Remove the pad from the element in release_pad. Fixes #364812.
+ * gst/playback/gstplaysink.c:
+ * tests/examples/seek/seek.c:
+ add framestepping to playbin2 and seek
-2006-10-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Avoid compiler warnings with -Wextra.
-2006-10-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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/sdp/gstsdpmessage.c:
+ sdp: Remove an unused variable.
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-30 14:17:41 +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), (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/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
- * tests/check/elements/subparse.c: (test_tmplayer_do_test),
- (GST_START_TEST), (subparse_suite):
- Add some basic unit tests for the above.
+2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-10-23 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-10-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
- * 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).
+2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.).
+2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@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/playback/gstplaysink.c:
+ playbin2: cosmetic change to avoid unnecessary line breaks
+ Looks nicer and works around gst-indent silliness.
-2006-10-20 Julien MOUTTE <julien@moutte.net>
+2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-2006-10-20 Michael Smith <msmith@fluendo.com>
+2009-05-28 15:21:42 +0100 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.
+ * 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>
-2006-10-20 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
- (plugin_init):
- Typefind mmsh header data packet to application/x-mmsh (#362625).
+2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/gdp/gstgdppay.c:
+ gdppay: set caps on outgoing buffers
+ Set caps on outgoing buffers because NULL caps confuse basetransform.
+ Fixes #583867
- * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
- (parse_subrip):
- Strip trailing newlines from subtitle text output.
+2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_change_state):
- Fix memleak; clear subparse->textbuf n state change function.
+2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraparse.c:
+ theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't require subrip (.srt) files to start with a chunk number of 1.
+2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-18 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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()
+2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * common:
+ Update common
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
- * tests/check/elements/playbin.c: (playbin_suite):
- Enable unit test for the above.
+2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
- * gst/playback/gstplaybin.c: (setup_sinks):
- The old pad activation spiel.
+2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: Add support for 16 bit grayscale in native endianness
- * 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).
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
-2006-10-17 Tim-Philipp Müller <tim at centricular dot net>
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
- * 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.
+2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-17 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * 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.
+2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
-2006-10-13 Julien MOUTTE <julien@moutte.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
- * 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.
+2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
-2006-10-13 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.spec.in:
+ Update spec file
- * 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.
+2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-10-13 Julien MOUTTE <julien@moutte.net>
+2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ basevideo: Fix memleak
-2006-10-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * ext/schroedinger/gstschrodec.c:
+ * ext/schroedinger/gstschroparse.c:
+ schro: Fix usage of adapter_masked_scan_uint32
+ Because *somebody* changed the API without telling me.
-2006-10-12 Julien MOUTTE <julien@moutte.net>
+2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
- * 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 !
+ * ext/schroedinger/gstschro.c:
+ schro: Change package name to GST_PACKAGE_NAME
-2006-10-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
- Patch by: Sebastien Cote <sebas642 at yahoo.ca>
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ basevideo: Add preset interface to encoder
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_finalize):
- Fix two small memory leaks (#361456).
+2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
-2006-10-10 Julien MOUTTE <julien@moutte.net>
+ * gst/audioresample/gstaudioresample.c:
+ Run liboil benchmark multiple times
+ The statistics function requires multiple runs, otherwise
+ it causes a divide by zero error.
- * tests/examples/seek/seek.c: (do_seek),
- (rate_spinbutton_changed_cb): When changing spinbutton we try
- to change the rate on the fly.
+2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-10 Wim Taymans <wim@fluendo.com>
+ * 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).
- * 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.
+2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * 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.
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2006-10-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
- Patch by: Ferenc Gerlits <fgerlits at gmail com>
+ * common:
+ Automatic update of common submodule
+ From d3a8fab to 888e0a2
- * gst/typefind/gsttypefindfunctions.c:
- Recognise XML files and XML-like files shorter than 256 bytes as
- well (fixes #359237).
+2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-10-09 Edgard Lima <edgard.lima@indt.org.br>
+ * win32/common/libgstvideo.def:
+ win32: Remove gst_adapter_masked_scan_uint32 from the exports
- 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>
+2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/xoverlay.c:
- (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
- Some more guards against invalid input.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: improve debug message
-2006-10-07 Julien MOUTTE <julien@moutte.net>
+2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
- * 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-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.
-2006-10-06 Stefan Kost <ensonic@users.sf.net>
+2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/xoverlay.c:
- Unref message in doc-example (spotted by Robert McQueen)
+ * 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.
-2006-10-06 Wim Taymans <wim@fluendo.com>
+2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- printf fix.
+ * win32/common/libgstvideo.def:
+ defs: add new symbols
-2006-10-06 Wim Taymans <wim@fluendo.com>
+2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * configure.ac:
+ autogen: pass -Wno-portability to automake to suppress warnings
+ GNU make is needed.
-2006-10-06 Michael Smith <msmith@fluendo.com>
+2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/floatcast/floatcast.h:
- Fix obviously-bogus macros; use the correct types.
+ * docs/libs/.gitignore:
+ gitignore: remove bogus *.sgml wildcard - these files are tracked in git
-2006-10-06 Wim Taymans <wim@fluendo.com>
+2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_change_state):
- Also call parent state change function to activate pads.
+ * gst/tcp/gsttcpclientsrc.c:
+ tcpclientsrc: this is not a live source
+ Don't mark us as a live source because we are not.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- Add some more debug info in mpeg typefinding.
+2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-10-06 Michael Smith <msmith@fluendo.com>
+ * 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.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Zero byte theora packets are valid and well-defined; don't warn on
- them.
+2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * 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
+2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-10-05 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ * tests/examples/seek/seek.c:
+ 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).
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
- Fix a simple mistake (see the docs)
- Fixes #359580
+2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin.c:
+ decodebin: use iterators instead of list
+ The list api is deprecated. Use threadsafe iterators instead.
- * 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.
+2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: configure caps on decodebin2
+ Implement the caps property by setting the configured caps on new decodebin2
+ objects.
+ Fixes #582749
- Patch by: James "Doc" Livingston <doclivingston at gmail com>
+2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some _caps_ref in some cases
+ Only mess with the caps refcount when we configure different caps.
- * 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.
+2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix potential caps leak
+ Free the user-configured caps in finalize.
- * 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.
+2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-30 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix typo in a debug statement.
+2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
- Move a recurring message to LOG level.
+2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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.
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- Removed empty * between paragraphs
+ * ext/theora/theoradec.c:
+ theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/README:
- Moved some documentation into .c file
+ * gst/adder/gstadder.c:
+ adder: add more logging and return value checking
-2006-09-29 Wim Taymans <wim@fluendo.com>
+2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c: (no_more_pads):
- Fix compilation.
+ * gst/adder/gstadder.c:
+ adder: handle the return value from iterator_fold
-2006-09-29 Wim Taymans <wim@fluendo.com>
+2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c: (new_caps):
- Remove g_print
+ * gst/adder/gstadder.c:
+ adder: use the pad in logging as objects
+ Helps to differenciate between source and sinks pads.
- * gst/playback/gstplaybin.c:
- Add some docs.
+2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: use parser for mp3 and rename variable
- * tests/check/Makefile.am:
- Re-enable cddabasesrc test to see if it works again
- now.
+2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst/playback/gstplaybasebin.c: (setup_subtitle),
- (gen_source_element):
- Handle invalid URIs a bit more gracefully.
+2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for v210 and v216 formats
- * tests/check/pipelines/oggmux.c:
- Remove obsolete comment.
+2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
-2006-09-29 Michael Smith <msmith@fluendo.com>
+ * 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
- * 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!
+2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
- * tests/check/pipelines/oggmux.c: (check_chain_final_state),
- (oggmux_suite):
- Reenable tests now that they pass.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add Y444, v210, v216 formats
-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).
+2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
-2006-09-28 Wim Taymans <wim@fluendo.com>
+ * 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/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-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>
-2006-09-28 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: add num-fds property
+ multifdsink::num-fds
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Set caps on outgoing buffers.
+2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
-2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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?)
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Implement Preset interface
-2006-09-28 Wim Taymans <wim@fluendo.com>
+2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/theora/theoraenc.c:
+ theoraenc: Implement Preset interface
-2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * ext/ogg/gstoggmux.c:
+ oggmux: Implement Preset interface
-2006-09-27 Wim Taymans <wim@fluendo.com>
+2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- Another attempt to make the gen64 buildbot happy.
-
-2006-09-27 Stefan Kost <ensonic@users.sf.net>
-
- 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
+ * gst/playback/gstplaysink.c:
+ playbin2: Fix cdda:// playback
+ Don't send async-start when the playsink has already been configured
+ before changing state.
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@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.
+ * configure.ac:
+ configure: require core CVS for gst_adapter_prev_timestamp()
+ which is used in the libvisual plugin.
-2006-09-27 Wim Taymans <wim@fluendo.com>
+2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * AUTHORS:
+ AUTHORS: fix my email
-2006-09-27 Wim Taymans <wim@fluendo.com>
+2009-04-22 18:35:40 +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/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 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- Temporarily disable test that fails on the bots for unknown reasons.
-
-2006-09-26 Philippe Kalaf <philippe.kalaf@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>
-
- * ChangeLog:
- ChangeLog surgery for missing bug-number
-
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Peter Kjellerstedt <pkj at axis com>
-
- * gst/playback/test.c:
- Fix compilation with uClibc and -Werror (#357591).
-
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- See if this makes the build bots happy.
-
- * tests/check/libs/cddabasesrc.c:
- UTF8-ise my name.
-
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
+ audioclock: make our internal time monotonic
+ Make the internal time increase monotonically.
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/Makefile.am:
- Need to build tag directory before cdda.
-
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
+ * ext/libvisual/visual.c:
+ visual: remove next_ts variable
+ We can remove the next_ts variable as we don't use it anymore.
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@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.
+ * 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.
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * 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.
- * 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.
+2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix misleading docs addition.
+2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/videotestsrc.c: (check_rgb_buf):
- Get rid of compiler warning the right way.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: seek to the requested byte offset, not the expected byte offset
-2006-09-22 Wim Taymans <wim@fluendo.com>
+2009-05-01 01:03:06 +0100 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
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: support more than just one channel
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- timestamps are uint.
+2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-22 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ propertyprobe: Fix typo in the docs
- * gst-libs/gst/interfaces/xoverlay.c:
- Remove unused statement from doc example.
+2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@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)
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theora.c:
+ * ext/vorbis/vorbis.c:
+ Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
-2006-09-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@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).
+ * 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.
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2009-04-28 11:37:14 +0200 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.
+ * ext/ogg/gstoggmux.c:
+ oggmux: small debug statement in DISCONT
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/libs/audio.c:
- Some more compilation fixes.
+ * 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>
-2006-09-21 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Early morning compilation fix.
+2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-20 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ rtpdepay: add some more comments
- * 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.
+2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: make sure values are ever increasing
- * 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.
+2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_get_times):
- change colorkey behaviour back according to #354773 comment 6/7
+2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-19 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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
+2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-19 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@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):
- Small cleanups, move error handling out of normal flow for clarity.
+ * gst-libs/gst/interfaces/tunernorm.h:
+ interfaces: Seperate some more struct definitions from typedefs
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/navigation.h:
* gst-libs/gst/interfaces/videoorientation.h:
- API: Add new interface to control video orientation (fixes #354908)
+ * gst-libs/gst/interfaces/xoverlay.h:
+ interfaces: Seperate some more struct definitions from typedefs
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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().
+ * win32/common/libgstinterfaces.def:
+ Add new functions to win32 exports
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
- Give better warning message (add object and detail).
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add new functions to the docs
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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-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.
-2006-09-18 Michael Smith <msmith@fluendo.com>
+2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/interfaces/colorbalance.c:
+ interfaces: Add docs for gst_color_balance_get_balance_type()
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
- * 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
+ * autogen.sh:
+ Run libtoolize before aclocal
+ This unbreaks the build in some cases. Fixes bug #582021
- * tests/examples/seek/seek.c: (main):
- add real meadi-buttons, add tool-tips for the seek-options, arrange
- seek options in a table
+2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-18 Michael Smith <msmith@fluendo.com>
+ * ext/pango/gsttextrender.c:
+ textrender: Correctly initialize the background for ARGB too
- * 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
+2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-18 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Michael Smith <msmith@fluendo.com>
+ * ext/pango/gsttextrender.c:
+ textrender: Don't always use ARGB if downstream supports it but take it's preference
- * 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.
+2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ textrender: Add support for ARGB and alignment properties
+ Fixes bug #581571.
- * 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.
+2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextrender.c:
+ textrender: Add ; after GST_BOILERPLATE to fix indention
- * 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.
+2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
- reviewed by: <delete if not using a buddy>
+2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
- * gst-libs/gst/audio/audio.h:
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
+ * 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.
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
- * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
- (gst_audio_convert_transform_caps):
- Get structure-name just once.
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Store cover art in vorbiscomments
+ Fixes bug #513373.
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c:
- Add docs about icydemux usage in connection with gnomevfssrc
+ * 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
- * 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.
+2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Fix wrong info in header due to copy & paste
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ Fix libdir for uninstalled gstreamer-app library
-2006-09-15 Wim Taymans <wim@fluendo.com>
+2009-05-12 01:59:01 +0100 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-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for APE tag caps
-2006-09-15 Wim Taymans <wim@fluendo.com>
+2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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 Wim Taymans <wim@fluendo.com>
+ configure: bump core requirement to last release
+ as that's more likely to be true than that we need
+ only 0.21.1.
- * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
- * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
- Use DEBUG_OBJECT more.
+2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.10.10 ===
-
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Michael Smith <msmith at fluendo dot com>
+ * common:
+ * configure.ac:
+ configure: rename CVS -> git in a couple of places
- * 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
+2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ configure: bump GLib requirement to GLib >= 2.16
+ as per the New Regime (see wiki).
- patch by: Wim Taymans <wim at fluendo dot com>
+2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/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.
-2006-09-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.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: support UYVY raw 4:2:2 in riff.
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/check/pipelines/oggmux.c: (oggmux_suite):
- Disable test that fails at the moment (killed after timeout).
+ * configure.ac:
+ Back to development -> 0.10.23.1
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
- Patch by: James Livingston <doclivingston at gmail.com>
+ * ext/theora/theoradec.c:
+ theoradec: fix buffer overrun on 422 decode.
- * 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.
+2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
-2006-09-04 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoradec.c:
+ theoradec: 444 support.
- patch by: Alessandro Dessina <alessandro nnva org>
+2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
- * ext/ogg/gstoggmux.c:
- Add cmml caps to oggmux. Fixes #353912
+ * ext/theora/theoradec.c:
+ theoradec: handle 422 images (as YUY2).
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
- * 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.
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theoradec: rearrange code in preparation for 422 and 444 support.
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+=== release 0.10.23 ===
- * 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.
+2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.23
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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").
+ * 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
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
- Fix stride for YVYU, should be word-aligned (#353658).
+ * 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
-2006-08-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_src_event):
- Fix build.
+ * 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.
-2006-08-31 Edward Hervey <edward@fluendo.com>
+2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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).
+ * 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
-2006-08-31 Edward Hervey <edward@fluendo.com>
+2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-08-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * gst/playback/gstplaysink.c:
+ playsink: initialize variable too
-2006-08-29 Wim Taymans <wim@fluendo.com>
+2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-08-29 Tim-Philipp Müller <tim at centricular dot 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:
+ 0.10.22.4 pre-release
- Patch by: Viktor Peters <viktor dot peters at gmail dot com>
+2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
- * 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.
+ * 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.
-2006-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstplaybin.c:
- Improve docs: add section about BUFFERING messages sent by playbin.
+ * 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
-2006-08-29 Michael Smith <msmith@fluendo.com>
+2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
- * 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.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "add can-activate-pull property to baseaudiosink"
+ This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
-2006-08-29 Edward Hervey <edward@fluendo.com>
+2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
- * 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.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "[baseaudiosink] add docs for can-activate-pull"
+ This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
-2006-08-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- fix buffer unreffing on a header push failure
+ [baseaudiosink] add docs for can-activate-pull
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
+ can-activate-pull.
-2006-08-28 Wim Taymans <wim@fluendo.com>
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
- (gst_audio_rate_chain):
- Make the metadata of the buffer writable before changing its
- flags.
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
-2006-08-28 Wim Taymans <wim@fluendo.com>
+2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-08-25 Michael Smith <msmith@fluendo.com>
+2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Fix spelling.
- Remove accidently included debug line.
+ * 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.
-2006-08-25 Wim Taymans <wim@fluendo.com>
+2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/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).
-2006-08-25 Michael Smith <msmith@fluendo.com>
+2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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:
+ 0.10.22.3 pre-release
- * 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.
+2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
-2006-08-25 Michael Smith <msmith@fluendo.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix unused variable in compilation with --disable-gst-debug
+ Fixes: #579668
- * 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.
+2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-08-24 Wim Taymans <wim@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From b3941ea to 6ab11d1
- * 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.
+2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-23 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * tests/check/elements/.cvsignore:
- make buildbot happy
+2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-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.
+ * 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.
-2006-08-23 Wim Taymans <wim@fluendo.com>
+2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- Waits for tasks to settle down so that we clean up correctly for
- valgrind.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.16 API, we require only 2.14
+ Fixes #579267.
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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-libs/gst/audio/gstbaseaudiosink.c:
+ 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
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
- * 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-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.
-2006-08-22 Wim Taymans <wim@fluendo.com>
+2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * configure.ac:
+ 0.10.22.2 pre-release
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * ChangeLog:
+ ChangeLog: regenerate changelog with the gen-changelog script
-2006-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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: Update po files from TP
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
- * tests/check/libs/tag.c: (GST_START_TEST):
- Some more tests for gst_tag_list_from_vorbiscomment_buffer().
+2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/libs/video.c:
+ check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
- * 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).
+2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-08-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * 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.
+2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/video/video.h:
+ video: Fix typo in the docs
- * 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.
+2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- Document GstNetBuffer.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for YVYU YUV colorspace
-2006-08-21 Stefan Kost <ensonic@users.sf.net>
+2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (GST_START_TEST),
- (audioconvert_suite):
- Add testcase for caps-size-explosion
+ * 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
-2006-08-20 Stefan Kost <ensonic@users.sf.net>
+2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_unit_size), (set_structure_widths):
- Lower debug, use g_assert in _get_unit_size
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ log: use G_GUINT64_FORMAT instead of llu
- * 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
+2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
-2006-08-18 Wim Taymans <wim@fluendo.com>
+ * 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
- * 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()
+2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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().
+2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
- * tests/check/libs/tag.c: (GST_START_TEST),
- (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
- More tests.
+ * ext/theora/theoraenc.c:
+ theoraenc: factor out keyframe forcing
+ See #578656
-2006-08-17 Wim Taymans <wim@fluendo.com>
+2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * AUTHORS:
+ * gst-libs/gst/fft/gstfft.c:
+ Give credit to Mark Borgerding (kissfft author)
+ and add myself to AUTHORS as well. Fixes #575638.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
- Fixed a bug in oggdemux debug code.
+2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
- * 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.
+ * 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.
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
- * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
- (test_buffer_timestamps), (cddabasesrc_suite):
- Add unit test for the above.
+2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
- * tests/check/Makefile.am:
- Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
- to see what happens.
+ * 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.
-2006-08-16 Wim Taymans <wim@fluendo.com>
+2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
- * 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).
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
+ Fixes #570768.
- * 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().
+2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
- Fix leak (#351502).
+2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-15 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
- * 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.
+2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
+ * 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()
- * 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).
+2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/playback/gstplaybin.c:
- Update max volume to the same value that the volume element uses.
+2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
-2006-08-14 Wim Taymans <wim@fluendo.com>
+ * 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.
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Less uglyness..
+2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-14 Wim Taymans <wim@fluendo.com>
+ * 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/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.
+2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-13 Wim Taymans <wim@fluendo.com>
+ * 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.
- * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
- (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
- Small code cleanup.
+2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
- * 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.
+2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-11 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playbin2: fix refcounting of visualisations
+ See #577794.
- * gst/adder/gstadder.c:
- Don't clip float values. Fixes #350900.
+2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-11 Andy Wingo <wingo@pobox.com>
+ * 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.
- * gst/tcp/gsttcp.c: Really fix the build?
+2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.h: For now, always disable deprecation here --
- fixes the build.
+ * tests/check/elements/appsink.c:
+ check: fix appsink test
+ Fix the appsink test now that the method signature changed.
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
- Float caps shouldn't have a "signed" field.
+ * gst/playback/gstplaybin2.c:
+ playbin2: handle missing input-selector
+ Gracefully degrade and disable stream selection when input-selector is
+ missing.
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
- * 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).
+ * 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.
-2006-08-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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.
-2006-08-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Sjoerd Simons <sjoerd at luon net>
+ * 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.
- * 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).
+2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-08-07 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.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).
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: don't leak pad-template
+ gst_element_class_add_pad_template() does not take ownership.
-2006-08-05 Andy Wingo <wingo@pobox.com>
+2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.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):
- API: GstTheoraParse::synchronization-points
- Add a property 'synchronization-points' to fix badly synchronized oggs.
+ * common:
+ Automatic update of common submodule
+ From d0ea89e to b3941ea
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/interfaces/navigation.c:
+ * sys/v4l/v4lsrc_calls.c:
+ navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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).
+ * ext/theora/theoradec.c:
+ theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
+ This fixes most seeking issues when used with gnonlin.
+ Fixes #543591
- * 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).
+2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From f8b3d91 to d0ea89e
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Recognise ancient RealAudio files (see #349779).
+2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't leak selector when getting current stream numbers.
- Patch by: Jens Granseuer <jensgr at gmx net>
+2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for Interplay's MVE format (#348973).
+ * 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
-2006-08-02 Wim Taymans <wim@fluendo.com>
+2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
- Patch by: Marcel Moreaux <marcelm at luon dot net>
+ * 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.
- * 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.
+2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-07-28 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * gst/audioresample/gstaudioresample.c: (audioresample_stop),
- (audioresample_set_caps):
- Don't leak references to the incoming caps. Clean them up when
- stopping.
+2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
- (gst_video_scale_finalize):
- Don't leak our temporary pixel buffer.
+ * tests/check/elements/.gitignore:
+ ignores: Ignore the videoscale check binary
- * tests/check/Makefile.am:
- * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
- (GST_START_TEST), (simple_launch_lines_suite):
+2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
- Fix leaks and re-enable the test for valgrind checking.
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_set_proxy to the win32 exports
-2006-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Sjoerd Simons <sjoerd at luon net>
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: don't forget to release locks in a few places
+ Might fix #576585.
- * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
- (plugin_init):
- Add typefind function for multipart/x-mixed-replace (#348916).
+2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-28 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't read over line ends when taking the last Cr or Cb
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_query_duration):
- Fix leak in duration query.
- Reflow some docs and notes.
+2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-28 Michael Smith <msmith@fluendo.com>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't write to few pixels and don't mix Cr and Cb
+ Fixes bug #577054.
- * 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.
+2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-28 Michael Smith <msmith@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.
- * 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.
+2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-07-27 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ docs: add a blurb about redirect messages to playbin2 docs
- patch by: Kai Vehmanen <kv2004 eca cx>
+2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@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_change_state):
- Don't send multiple newsegments with different formats.
- Fixes #348677.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix little typo in the comments
-2006-07-26 Wim Taymans <wim@fluendo.com>
+2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * 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.
-2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ docs: fix typo in gst_rtsp_message_unset() API docs
-2006-07-24 Wim Taymans <wim@fluendo.com>
+2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Don't try to align a sample to an unknown value.
+ * 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.
-2006-07-24 Wim Taymans <wim@fluendo.com>
+2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
+ This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
-2006-07-24 Wim Taymans <wim@fluendo.com>
+2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- Don't try to calculate silence samples, base class does this much
- better now.
+ * 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-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.
+2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstringbuffer.h:
- Add _CAST macro.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: reset whole message (was sizeof pointer instead of sizeof type)
-2006-07-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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-libs/gst/interfaces/mixer.c:
+ doc: Fix a typo in the GstMixer docs
-2006-07-21 Andy Wingo <wingo@pobox.com>
+2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
- test to the one in vorbisenc. Also commented out.
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix linear scaling for one byte components
+ Fixes bug #577054.
- * tests/check/pipelines/vorbisenc.c:
- (test_discontinuity): New test, commented out until Mike lands
- some elite vorbisenc patches.
+2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
- Bufferstraw was actually factored out of these tests. Now we share
- code yay.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Fix 4tap scaling of YUYV and friends
- * configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
- for bufferstraw addition to gstcheck.
+2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-21 Wim Taymans <wim@fluendo.com>
+ * 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.
- * ext/theora/theoradec.c: (clip_buffer):
- Better clipping.
+2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-21 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/videoscale.c:
+ videoscale: Add some more unit tests
- * 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.
+2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
- Fix mem leak.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use bilinear instead of 4tap scaling for heights < 4
+ Partially fixes bug #577054.
-2006-07-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2006-07-20 Wim Taymans <wim@fluendo.com>
+2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (cb_newpad), (main):
- Example of a small audio/video player using decodebin.
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
+ Partially fixes bug #577054.
-2006-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- Add 'fact' chunk id
+ * 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).
-2006-07-19 Wim Taymans <wim@fluendo.com>
+2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/check/Makefile.am:
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a lot of unit tests
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Add comment about better default MTU size.
+2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- Small cleanups, start docs.
+ * gst/videoscale/gstvideoscale.c:
+ videocale: Add support for video/x-raw-gray with bpp=depth=8
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Martin Szulecki
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
- * 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).
+2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
- * gst-libs/gst/tag/gstid3tag.c:
- Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
+2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gsttageditingprivate.h:
- * gst-libs/gst/tag/gstvorbistag.c:
- Some more random const-ifications.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
-2006-07-18 Stefan Kost <ensonic@users.sf.net>
+2009-03-27 19:09:47 +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_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/videoscale/gstvideoscale.c:
+ videoscale: Add support for v308 YUV colorspace
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_fixate_channel_positions):
- Const-ify two arrays.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Add my copyright to the 4tap scalers
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Enable 4-tap scaling for all supported formats
-2006-07-17 Wim Taymans <wim@fluendo.com>
+2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB565 and RGB555
-2006-07-14 Wim Taymans <wim@fluendo.com>
+2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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>
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for UYVY
- * configure.ac:
- Bump nano back to CVS
+2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-=== release 0.10.9 ===
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for YUY2 and YVYU
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- releasing 0.10.9, "I walk the line"
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB and BGR
-2006-07-14 Michael Smith <msmith@fluendo.com>
+2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
+
+2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Fix drawing of UYVY text borders
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Fix missing g_unlock from the previous commit
+2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Add support for UYVY colorspace
+ Fixes bug #378094.
- * 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.
+2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: do some more cleanup
+ Free the groups when we go to READY.
+ Allow for NO_PREROLL elements.
- * gst/playback/gstdecodebin.c: (find_compatibles):
- Fix a caps leak when linking (#347304)
+2009-03-25 16:37:28 +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)
+ * 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.
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-03-25 16:35:22 +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.
+ * 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.
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
- Reset the decoder in finalize so that all fields get cleared.
+ * 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
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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.
+ audiosrc: improve 'Dropped n samples' warning message
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@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.
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: add more support for subpictures
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- basesink calculates silence sample correctly for us.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: first support for subpictures
+ Add beginnings of subpicture support.
-2006-07-12 Wim Taymans <wim@fluendo.com>
+2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * tests/examples/seek/seek.c:
+ seek: print tags from the different tracks
-2006-07-12 Edward Hervey <edward@fluendo.com>
+2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@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
+ * 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.
-2006-07-11 Jan Schmidt <thaytan@mad.scientist.com>
+2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-2006-07-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/gstplaysink.c:
+ playbin2: rescan audio sinks for volume/mute
+ Rescan the audio sinks for the mute and volume properties.
+ fixes #576180.
-2006-07-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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
-2006-07-08 Andy Wingo <wingo@pobox.com>
+2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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
-2006-07-06 Wim Taymans <wim@fluendo.com>
+2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@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)
+ * 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/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).
+2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-06 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/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.
+2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-07-06 Tim-Philipp Müller <tim at centricular dot net>
+ * 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).
- * 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.
+2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-07-05 Sebastien Moutte <sebastien@moutte.net>
+ * gst/videoscale/gstvideoscale.c:
+ indent: fix damange that gst-indent did some time ago
- * win32/common/libgsttag.def:
- Export some new functions.
- * win32/vs6/libgstogg.dsp:
- Add a link to libgsttag-0.10.lib.
+2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-04 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Some const-ification.
+2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-04 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: reset counter
+ reset the number of pending dynamic operations back to 0 when we reuse
+ uridecodebin.
+ Fixes #576190
- * 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.
+2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-07-03 Edward Hervey <edward@fluendo.com>
+ * 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.
- * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
- Remove unused variable.
+2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
-2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Set caps on non-header ouput buffers.
+ Fixes #576142.
- * Makefile.am:
- include lcov.mak
- * configure.ac:
- add GCOV_LIBS to GST_LIBS
+2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-07-02 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: Add some more debug
+ Add some more info about the selected streams.
- Patch by: Michael Sheldon <webmaster at mikeasoft com>
+2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsasrc.c:
- Add 32 bps to template caps and increase channels range
- from [1,2] to [1,MAX]. See #346326.
+ * 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.
-2006-06-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
- * 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
+ * gst/playback/gstqueue2.c:
+ win32: fix seeking in large files
+ Fix Seeking in large files by using the 64-bit seek functions.
+ Fixes #576019
-2006-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
- (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
- fix logging
+ * 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.
-2006-06-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: more cleanups and docs.
+ Add some more comments and use g_list_prepend().
- When going from READY to PAUSED, restore the fakesink, so that
- it is there when decodebin gets reused.
+2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-23 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/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).
+2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/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.
+2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-22 Wim Taymans <wim@fluendo.com>
+ * tests/icles/playbin-text.c:
+ icles: print better error and warning messages
+ --
- * 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.
+2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst-libs/gst/tag/tag.h:
- Oops, make that 'Since: 0.10.9'.
+2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Cody Russell <bratsche at gnome org>
+2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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.
-2006-06-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-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
+2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp-marshal.list:
- Added new marshaller code for the new signal.
+ * tests/icles/playbin-text.c:
+ example: use appsink instead of fakesink
+ Use appsink instead of fakesink to get the subtitles.
+ Make things more pretty.
- * tests/check/elements/multifdsink.c: (GST_START_TEST),
- (multifdsink_suite):
- Added testcases for new burst methods.
+2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-19 Edward Hervey <edward@fluendo.com>
+ * 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/theora/theoradec.c: (clip_buffer), (theora_dec_push):
- Implement clipping for accurate seeking.
- Closes #345225
+2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-19 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/appsink.c:
+ tests: fix include in the appsink test
+ Fix dist by doing the right include.
- Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
+2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
- (gst_video_scale_transform):
- Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
+ * 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.
-2006-06-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Fix --disable-external (can't set conditionals conditionally,
- #343602).
+ * 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.
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/audioresample.c: (test_reuse),
- (audioresample_suite):
- Add test case for bug #342789 fixed below.
+ * 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).
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
- * tests/check/elements/audioresample.c: (setup_audioresample),
- (cleanup_audioresample):
- Need to set element state here so that ::start and ::stop are
- called.
+2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-06-16 Wim Taymans <wim@fluendo.com>
+ * 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.
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.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.
+ * tests/check/Makefile.am:
+ check: Ignore alsamixer in the states test too
-2006-06-16 Wim Taymans <wim@fluendo.com>
+2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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
+ * sys/v4l/v4l_calls.c:
+ v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
- Must dereference pointer to fourcc in the debug statement.
+2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-16 Stefan Kost <ensonic@users.sf.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.
- * 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
+2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-15 Wim Taymans <wim@fluendo.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add some debug info
+ Add some debug info to log when the seek worked.
- * 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.
+2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/playback/gststreaminfo.c: (cb_probe):
- Try GST_TAG_CODEC as fallback when extracting the
- codec name; more debug info.
+2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
- * 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).
+2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-06-14 Wim Taymans <wim@fluendo.com>
+ * ext/theora/theoradec.c:
+ theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
- Patch by: Alessandro Decina <alessandro at nnva dot org>
+2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/playback/gsturidecodebin.c:
+ playbin2: fix raw elements like cdda://
+ Fix a fixme with a one liner and make cd playback work again.
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- Add IDs for 'bext' chunks (see #343837).
+ * 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.
-2006-06-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * 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/subparse/samiparse.c: (sami_context_pop_state),
- (handle_start_font), (end_sami_element):
- Honour font face tags in SAMI subtitles (#344503).
+2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst-plugins-base.spec.in:
+ Spec: fix up deps
- * po/POTFILES.in:
- add missing files containing translatable strings
+2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix parsing of the timeout parameter
+ --
- * docs/libs/tmpl/.cvsignore:
- we don't want those *.sgml files in CVS either
+2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: fix g_return condition
+ when parsing a data message, we require a data message.
- * docs/libs/.cvsignore:
- * tests/check/elements/.cvsignore:
- * tests/check/libs/.cvsignore:
- ignore more
+2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-06-11 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.
- * docs/libs/Makefile.am:
- also commiting the changed Makefile.am (added more libs to the
- doc-build)
+2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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
+2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * configure.ac:
- use GST_PKG_CHECK_MODULES, cleans up output
+2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ adder: log details in getcaps like in setcaps
- * 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.
+2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/MANIFEST:
+ win32: update MANIFEST, fixing 'make dist'
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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>
+ * common:
+ Automatic update of common submodule
+ From 7032163 to f8b3d91
- * configure.ac:
- back to TRUNK
+2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
-=== release 0.10.8 ===
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add photoshop typefind functions
+ Add photoshop typefind functions.
+ Fixes #574516.
-2006-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.10.8, "Moar gij ziet mij nie"
+ * 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.
-2006-06-07 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@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
-
-2006-06-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-06-06 Michael Smith <msmith@fluendo.com>
-
- * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
- (theora_parse_drain_queue):
- Mark DELTA_UNIT on non-keyframes.
-
-2006-06-03 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Michael Smith <msmith@fluendo.com>
-
- * 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-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.
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@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-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: use gstreamer dump mem
+ --
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@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-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: use glib base64 encoder
+ --
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@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/gstdecodebin2.c:
+ Unblock blocked ghostpads when shutting down. Fixes #574293.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/audioconvert/audioconvert.c: (float):
- More correct float->int conversion.
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add mapping for Fraps video codec.
+ Found through insanity testrun. Confirmed mapping in libavformat.
-2006-06-02 Michael Smith <msmith@fluendo.com>
+2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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-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.
-2006-06-02 Wim Taymans <wim@fluendo.com>
+2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Use the proper data pointer instead of poking random memory.
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
- * 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
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix compilation on windows.
+ Remove unused variable when building for windows.
+ Fixes #574443.
-2006-06-01 Stefan Kost <ensonic@users.sf.net>
+2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * common:
+ Automatic update of common submodule
+ From ffa738d to 7032163
-2006-06-01 Stefan Kost <ensonic@users.sf.net>
+2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * common:
+ Automatic update of common submodule
+ From 3f13e4e to ffa738d
-2006-05-31 Wim Taymans <wim@fluendo.com>
+2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 3c7456b to 3f13e4e
-2006-05-31 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/Makefile.am:
- set a magic variable to indicate we know the docs are incomplete
+ * common:
+ Automatic update of common submodule
+ From 57c83f2 to 3c7456b
-2006-05-30 Sebastien Moutte <sebastien@moutte.net>
+2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * win32/common/libgstvideo.def:
- export gst_video_calculate_display_ratio
- * win32/vs6/libgstvideoscale.dsp:
- add link to libgstvideo-0.10.lib
+ * 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.
-2006-05-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add theora mapping
+ Add theora mappings. See #574169.
-2006-05-30 Wim Taymans <wim@fluendo.com>
+2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
- (gst_text_overlay_text_event):
- Added some FIXMEs.
+ * 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()
-2006-05-30 Wim Taymans <wim@fluendo.com>
+2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * Makefile.am:
+ * win32/common/audio-enumtypes.c:
+ win32: indent copied *-enumtypes.c files in make win32-update
- * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
- Added check for release_pad in adder.
+2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-30 Wim Taymans <wim@fluendo.com>
+ * win32/MANIFEST:
+ win32: update MANIFEST
- * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
- Fix build again.
+2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * win32/common/config.h:
+ win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
- * 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
+2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-30 Michael Smith <msmith@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).
- * 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.
+2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-30 Wim Taymans <wim@fluendo.com>
+ * .gitignore:
+ gitignore: ignore more
- * 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.
+2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
-2006-05-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fix build on Mac OS X
- * gst/adder/gstadder.c: (gst_adder_request_new_pad):
- Don't leak pad name.
+2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-05-29 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/adder/gstadder.c: (gst_adder_query_duration),
- (forward_event_func), (forward_event), (gst_adder_src_event):
- Fix adder seeking.
- Make query/seeking code threadsafe.
+2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ rtsp: fix parsing of 'now-' ranges.
+ --
-2006-05-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * 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.
- * 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).
+2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: add appsrc symbols to standard section
+ --
-2006-05-29 Michael Smith <msmith@fluendo.com>
+2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/adder/gstadder.c:
+ adder: add variants for unsigned to fix warnings for unneeded check
+ For unsigned int out+in can't be < 0.
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/check/elements/volume.c:
- make it compile again
+ * gst/subparse/gstsubparse.c:
+ subparse: use the right variable in debug log, encoding is not yet initialized
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
- * tests/check/elements/volume.c: (volume_suite):
- disable test until #343196 gets resolved
+ * sys/v4l/v4l_calls.c:
+ v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
+2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: add missing break in event handling, remove dead code
-2006-05-28 Edward Hervey <edward@fluendo.com>
+2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-05-27 Edward Hervey <edward@fluendo.com>
+2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-05-27 Edward Hervey <edward@fluendo.com>
+2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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>
-2006-05-27 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * README:
- Replace current README (containing the release notes from
- some 0.9.x version) with a proper README taken from the core.
+2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@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.
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: remove , from last enum member
+ Remove , from last enum member to improve compatibility with other compilers.
- * configure.ac:
- Bump requirements to core CVS (needed for vorbis
- typefinding to work).
+2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
-2006-05-24 Edward Hervey <edward@fluendo.com>
+ * 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/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.
+2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-23 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: remove debugging g_message
+ --
- * 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
+2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
+ * 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()
- * 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.
+2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/tcp/Makefile.am:
- fdstresstest doesn't need Gtk+, fix compilation if
- gtk is not available (#342566).
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_get_url definition
+ No, I'm not wim's buildslave, seriously.
-2006-05-19 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- 80 line columns
- Removed redundant floor()
+2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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()
- * 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.
+2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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-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.
+2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
-2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ Bump glib requirement to 2.14
- * 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)
+2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
-2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/gio/gstgiobasesink.c:
+ gio: Use correct format modifier for size_t
+ Fixes bug #573528.
- * 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.
+2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
-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-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Use correct types for some functions on Win32
+ Fixes bug #573529.
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix warning about using unitialized value.
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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-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
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/gstalsa.c:
+ alsa: release pcminfo after the strdup
- * 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>
+2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/subparse/samiparse.c: (handle_start_font):
- Don't ignore return value of strtol (++compiler_happiness).
+2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-17 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Young-Ho Cha <ganadist chollian net>
+2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.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/gstsubparse.c:
+ subparse: don't leak line, if flushing
- * gst/subparse/samiparse.c: (characters_sami):
- Output is pango markup, so we need to escape text
- between tags (#342143).
+2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiosink.c:
+ giosink: reflow error handling to not leak uri
- * 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.
+2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: remove unused code/variables
- * 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.
+2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/ximage/ximagesink.c:
+ ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
- * 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.
+2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@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/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>
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- Query supported sample rates. Fixes #341732.
+2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-05-15 Julien MOUTTE <julien@moutte.net>
+ * win32/common/libgstapp.def:
+ win32: Add new libgstapp symbol
- * 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.
+2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-15 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybasebin.c:
+ docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
- * 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>
+2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't use libxml functions in the typefinding code.
+ * 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
-2006-05-15 Wim Taymans <wim@fluendo.com>
+2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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)
+ * gst-libs/gst/video/video.h:
+ docs: fix newly added interlace constants and plug holes in video format docs
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
+
+2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/ximage/ximagesink.c:
+ ximagsink: do not access uninitialized height variable.
+ Exit like in xvimagesink, if we have partial caps.
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
+ * Makefile.am:
* 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>
+ * 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.
+
+2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fixed a typo.
+2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst/tcp/gstmultifdsink.c:
+ rtsp, multifdsink: Unify the use of union gst_sockaddr.
+
+2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
+
+ * common:
* configure.ac:
- * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
- (plugin_init):
- Add tentative support for libvisual-0.4 (#336881).
+ build: Update shave init statement for changes in common. Bump common.
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_font):
- Need to map "silver" colour explicitly (#169936).
+ * 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.
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * 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.
- * 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).
+2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-05-14 Michael Smith <msmith@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 9cf8c9b to a6ce5c6
- * 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.
+2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.10.7 ===
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
+ Fixes: #565777
-2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.10.7, "Leave the gun"
+ * 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.
-2006-05-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-25 11:07:20 +0100 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.
+ * tests/icles/stress-playbin.c:
+ Print the errors more clearly
+ Print some more verbose messages when dealing with errors.
-2006-05-12 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-25 10:08:29 +0100 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/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.
-2006-05-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-25 10:05:38 +0100 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/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.
-2006-05-12 Sebastien Moutte <sebastien@moutte.net>
+2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
- * win32/MANIFEST:
- update win32 files listing
+ * gst/playback/gstplaybin2.c:
+ Playbin2: Don't leave unused parameters in debug statements.
+ Fixes build on macosx
-2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
- patch by: Sjoerd Simons (sjoerd@luon.net)
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
- * 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
+2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-11 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Add some G_UNLIKELY because we can
+ Add a G_UNLIKELY when checking the shutdown variable.
- * 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.
+2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
- * 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-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
-2006-05-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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/tcp/gstmultifdsink.c:
+ multifdsink: Fix strict aliasing error using a union
-2006-05-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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-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.
-2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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
+ * 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.
-2006-05-11 Michael Smith <msmith@fluendo.com>
+2009-02-24 14:36:39 +0100 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.
+ * 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
-2006-05-11 Wim Taymans <wim@fluendo.com>
+2009-02-24 14:06:38 +0100 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/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add conversion from/to YVYU colorspace
+ Fixes bug #572872.
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
- * tests/check/elements/multifdsink.c: (wait_bytes_served),
- (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
- add two more tests, one doing streamheader
+ * 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.
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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/gstplaybin2.c:
+ playbin2: fix deadlock when shutting down. Fixes #572577.
-2006-05-10 Wim Taymans <wim@fluendo.com>
+2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ * tests/icles/stress-playbin.c:
+ stress-playbin: make more flexible, e.g. also useful for playbin2
+
+2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.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-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>
+
+2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
- Require core CVS for the new newsegment stuff.
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Use shave for the build output
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: Sjoerd Simons <sjoerd at luon net>
+ * win32/common/libgstrtsp.def:
+ win32: Add new symbol to libgstrtsp.def
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
- Register nick for enum value (#341160).
+2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-09 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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
+2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Wim Taymans <wim@fluendo.com>
+ * tests/check/libs/netbuffer.c:
+ Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
- * 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.
+2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Edward Hervey <edward@fluendo.com>
+ * ext/theora/theoraparse.c:
+ theoraparse: Use the correct unref functions
- * tests/icles/stress-xoverlay.c:
- Fix if core was built without parsing support.
+2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add SEDG (Samsung MPEG-4) fourcc.
+2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-05-09 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Unref the actual buffer instead of the memory address of the buffer
- * tests/icles/stress-xoverlay.c:
- * tests/examples/volume/volume.c:
- Fix if core was built without parsing support.
+2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
- * tests/examples/seek/seek.c:
- Disable the parse_launch example if core was built without parsing
- support.
+ * common:
+ Automatic update of common submodule
+ From 5d7c9cc to 9cf8c9b
-2006-05-08 Edward Hervey <edward@fluendo.com>
+2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
- * autogen.sh: (CONFIGURE_DEF_OPT):
- libtoolize on Darwin/MacOSX is called glibtoolize
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgstvideo.def:
+ win32/common: Update .def files for recent API addition
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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
+ * tests/check/libs/rtp.c:
+ tests: Fix indentation
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
- (adder_suite):
- Shuffle NULL state change around and raise timeout more
+ * 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.
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
- * 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
+ * common:
+ Automatic update of common submodule
+ From 80c627d to 5d7c9cc
- * tests/check/elements/adder.c: (adder_suite):
- Raise timeout to make buildbot happy
+2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-07 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.
- * 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
+2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-05 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/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.
+2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-05 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
-2006-05-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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>
- * 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.
+2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-05-05 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Fix 'Since' tags
- * gst/playback/gstdecodebin.c: (cleanup_decodebin):
- Properly remove ghostpads. Fixes #340392
+2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-05-04 David Schleef <ds@schleef.org>
+ * 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/typefind/gsttypefindfunctions.c:
+2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-03 Jan Schmidt <thaytan@mad.scientist.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/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg_ts_probe_headers), (mpeg_ts_type_find):
+2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
- 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/riff/riff-media.c:
+ Add some more mappings for h264 in riff
- Make the probability for detecting MPEG Transport Streams based on the
- number of sequential headers we successfully detected.
+2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-03 Wim Taymans <wim@fluendo.com>
+ * win32/common/libgstrtsp.def:
+ Add new RTSP symbols to def files
+ Add the new RTSP symbols to the windows def file.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
- (vorbis_dec_push), (vorbis_dec_chain):
- Still produce an error when we receive an empty packet.
+2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-03 Wim Taymans <wim@fluendo.com>
+ * 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>
- * 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.
+ * 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()
- * 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.
+2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-05-03 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:
+ 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>
- * 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.
+ * gst/audioresample/gstaudioresample.c:
+ 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.
-2006-05-02 Stefan Kost <ensonic@users.sf.net>
+2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/videotestsrc/videotestsrc.c:
+ ffmpegcolorspace/videotestsrc: Use v308 instead of V308
-2006-05-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2006-04-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Protect against unexpected NULL strf_data buffer.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
+ Partially fixes bug #571147.
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * 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).
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- when a test fails, give an indication of which it is
+ * 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.
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- compile fix; strtol() needs <stdlib.h>
+ * .gitignore:
+ Make git ignore backup files.
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
- * 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
-
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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
+2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
- (alsasink_parse_spec):
- query width capabilities from alsa, fixes #338919
+2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-28 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.
- * 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
+2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
-2006-04-28 Michael Smith <msmith@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/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.
+2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
+ Partially fixes bug #571147.
- * 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.
+2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
-2006-04-28 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/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
+2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
-2006-04-28 Wim Taymans <wim@fluendo.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>
- * 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.
+ * 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.
-2006-04-28 Wim Taymans <wim@fluendo.com>
+2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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/gstaudiosrc.c: (gst_audio_src_init):
- Don't set live to TRUE as this is the default in the parentclass.
+2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-28 Wim Taymans <wim@fluendo.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.
- * 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
+2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
+ The flags where present but actually not been taken into account.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- Handle 0/1 framerate correctly Fixes #331901.
+2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * gst/audioresample/gstaudioresample.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.
- * tests/check/elements/audioconvert.c: (get_float_caps),
- (GST_START_TEST), (audioconvert_suite):
- Added check for correct clipping when doing float samples
- in audioconvert.
+2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst/videorate/gstvideorate.c: (gst_video_rate_event),
- (gst_video_rate_chain):
- Print more debugging info.
+2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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
+2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-28 Wim Taymans <wim@fluendo.com>
+ * 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/audioconvert/audioconvert.c: (float):
- correctly clip float samples > 1.0. Fixes #338718
+2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstrtsp.def:
+ Add new methods
+ Add new methods to the windows def file.
- Patch by: Young-Ho Cha <ganadist at chollian net>
+2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
-2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
* configure.ac:
- back to HEAD
+ * gst/audioresample/gstaudioresample.c:
+ Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
-=== release 0.10.6 ===
+2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * configure.ac:
- releasing 0.10.6, "Chao"
+2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-26 David Schleef <ds@schleef.org>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ Fix documentation for autoplug-select
+ fix the documentation strings for the autoplug-select signal.
+ Fixes #570142.
- * gst/videoscale/gstvideoscale.c: Add call to oil_init().
- Fixes #338897.
+2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * configure.ac:
- * win32/common/config.h:
- new prerelease
+2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ Its "Base Library" and not just "Library".
- patch by: Wim Taymans
+2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ Link to the class, as we can't link to the members yet.
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
- Patch by: Edward Hervey <edward@fluendo.com>
+ * 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/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.
+2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
+2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- fix typefinding on some ISO files. Fixes #339212.
+ * gst/playback/gstplaybin2.c:
+ Unref source in playbin2's finalize method
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * 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/riff/riff-media.c: (gst_riff_create_video_caps):
- add another H264 fourcc. Fixes #339047.
+2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- Patch by: Jan Schmidt
+2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_bufferalloc):
- Restore old StreamSelector behaviour.
- Fixes #338419.
+ * gst/playback/gstplaybin2.c:
+ Unref peer request pad after releasing it, since we hold a reference.
-2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
- * configure.ac:
- * win32/common/config.h:
- prerelease
+ * gst/playback/gstplaybin2.c:
+ Fix caps leak in playbin2.
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/playback/gstplaybin2.c:
+ Unref active pad from selector when finding active stream.
- * 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.
+2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-11 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Free uris when finalizing playbin2 instance.
- * 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).
+2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ Unref pads when iterating over them in analyse_source.
+ Fixes leak of source's srcpad when using uridecodebin.
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_get_property):
- Don't leak pad name.
+2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
-2006-04-10 Michael Smith <msmith@fluendo.com>
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ Add releaseinfo with online url.
- * 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.
+2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybasebin.c:
+ Fix compilation warning on Forte
- * 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.
+2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ * gst/adder/gstadder.c:
+ Don't do void pointer arithmetic.
- * 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.
+2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ * common:
+ Bump common
- * 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.
+2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-04-10 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ * common:
+ Use a symbolic link for the pre-commit client-side hook
- * ext/theora/theoradec.c: (theora_dec_src_event),
- (theora_handle_data_packet):
- Some more debug info.
+2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
- * tests/examples/seek/seek.c: (start_seek), (main):
- Print element messages too.
+ * .gitignore:
+ Add more files/directories to ignore
-2006-04-09 Sebastien Moutte <sebastien@moutte.net>
+2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * 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.
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
+2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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>
+ * 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()
- * 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
+2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
- More debug to trace why my USB headset is not working with gst
+2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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()
- * 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.
+2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
-2006-04-07 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).
- * 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.
+2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-04-07 Michael Smith <msmith@fluendo.com>
+ * ext/gio/Makefile.am:
+ Fix include order for gio plugin
- * 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.
+2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-04-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * win32/common/config.h:
+ Update win32 config.h for 0.10.22.1 dev cycle
- * 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.
+2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.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)
+ * .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
-2006-04-06 Andy Wingo <wingo@pobox.com>
+2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
- * 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.
+2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
- buffers being pushed out. Fixes oggmux ! multifdsink.
+ * 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.
-2006-04-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@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/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.
-2006-04-04 Andy Wingo <wingo@pobox.com>
+2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@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/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.
- * 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.
+2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-03 Wim Taymans <wim@fluendo.com>
+ * 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
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (theora_dec_src_event):
- Don't try to store a gdouble in a gboolean.
- Small cleanups.
+2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-04-03 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
-
-2006-04-03 Andy Wingo <wingo@pobox.com>
+2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
- a g_print.
-
- * ext/theora/theora.c (plugin_init): Register theoraparse.
+ * ext/alsa/gstalsasink.c:
+ Improve debug message
+ Improve the debug message when alsa returns an error.
- * ext/theora/gsttheoraparse.h:
- * ext/theora/theoraparse.c: New files implementing a theora
- parser. Now we can properly remux ogg/theora+vorbis, yay.
+2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
+ * 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
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Fixed the sample pipeline (see #323798)
+ * gst/typefind/gsttypefindfunctions.c:
+ Add typefinder for Mobile XMF. Fixes bug #568707.
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
* configure.ac:
- * win32/common/config.h:
- * win32/common/config.h.in:
- use AS_VERSION and AS_NANO
- more cleanups
+ 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.
-2006-03-31 Andy Wingo <wingo@pobox.com>
+2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * gst/playback/gstplaybasebin.c:
+ Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
- * 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.
+2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
- and keep track of data needed for deriving granulepos and
- timestamps for buffers.
+ * common:
+ Update common snapshot.
-2006-03-29 Wim Taymans <wim@fluendo.com>
+2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
- Patch by: j^ <j at bootlab dot org>
+ * common:
+ Fix pre-commit hook
- * 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
+2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-03-29 Wim Taymans <wim@fluendo.com>
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
- * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
- (stop_seek):
- Don't let double and triple clicks mess up our state.
+2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ Back to devel -> 0.10.22.1
- * 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).
+2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
+ * autogen.sh:
+ * common:
+ Install and use pre-commit indentation hook from common
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
- (gst_gnome_vfs_src_get_icy_metadata):
- Fix some minor memory leaks (#336194).
+ * 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.
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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).
+ * autogen.sh:
+ autogen.sh : Use git submodule
-2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+=== release 0.10.22 ===
- * 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 ?
+2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.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:
- update
+ Release 0.10.22
+ Original commit message from CVS:
+ Release 0.10.22
-2006-03-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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).
+ * 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).
-2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * configure.ac:
- update core requirement to 0.10.4.1 because of async_playback
- vmethod on GstBaseSink
+2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
+ 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
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
- * gst/adder/gstadder.c: (gst_adder_init):
- use DEBUG_FUNCPTR for collectpads
+2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * 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
+2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-23 Sebastien Moutte <sebastien@moutte.net>
+ 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.
- * 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>
+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.
- * 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...
+2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-23 Wim Taymans <wim@fluendo.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.
- * 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.
+2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- Enable QoS with new method in base class.
+ 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.
-2006-03-23 Wim Taymans <wim@fluendo.com>
+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>
- Patch by: Julien MOUTTE <julien at moutte dot net>
+ * 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.
- * 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.
+2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
-2006-03-23 Wim Taymans <wim@fluendo.com>
+ 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.
- Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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-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.
-2006-03-22 Edward Hervey <edward@fluendo.com>
+2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
- * docs/design/design-decodebin.txt:
- Added design document for new decodebin
- (Target Caps): text/x-pango-markup is also a default target caps.
+ 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.
-2006-03-22 Wim Taymans <wim@fluendo.com>
+2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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-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.
-2006-03-22 Wim Taymans <wim@fluendo.com>
+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'.
- * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
- (stop_seek), (scrub_toggle_cb), (main):
- Add scrub checkbox.
+2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-21 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
- (gst_ogg_parse_chain):
- Fix very inefficient usage of linked lists (#335365).
+2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
-2006-03-21 Edward Hervey <edward@fluendo.com>
+ 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.
- * 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*.
+2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
-2006-03-20 Julien MOUTTE <julien@moutte.net>
+ 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/xvimage/xvimagesink.c:
- (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
- by Jan Schmidt.
+2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
-2006-03-19 Tim-Philipp Müller <tim at centricular dot net>
+ 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/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>
+2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2006-03-17 Jan Schmidt <thaytan@mad.scientist.com>
+2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- * win32/common/config.h:
- Revert accidental commits to these files.
+ 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.
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
+2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
- Patch by: Michal Benes <michal dot benes at xeris dot cz>
+ 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).
- * tests/Makefile.am:
- Don't try to build tests in tests/icles if we
- don't have X (#323852)
+2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst-libs/gst/tag/gstid3tag.c:
- Add TXXX frame identifiers for replaygain stuff as used
- by some taggers (see #323721).
+2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-03-16 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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.
+2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst-libs/gst/tag/gstid3tag.c:
- Don't crash on unknown ID3v2 TXXX frames.
-
-2006-03-15 Jan Schmidt <thaytan@mad.scientist.com>
+2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
- Chain up to the parent finalize method.
- Add 32-bit sample size to the template caps.
+ 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.
- * 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.
+2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@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/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.
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+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().
- 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).
+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.
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
+2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Patch by: Sebastien Moutte <sebastien moutte 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/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.
+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>
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+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.
- * 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>
+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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for 24bpp DIB (#305279).
+2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
-2006-03-14 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * configure.ac:
- back to CVS.
+2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
-=== release 0.10.5 ===
+ * 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>
-2006-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
- * configure.ac:
- releasing 0.10.5, "Net"
+2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-13 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * docs/plugins/Makefile.am:
- Part of previous cdparanoiasrc docs fixes, forgot to commit.
-
-2006-03-12 Tim-Philipp Müller <tim at centricular dot net>
+2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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/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.
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- More GstCddaBaseSrc docs.
+2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-03-12 Tim-Philipp Müller <tim at centricular dot 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.
- * 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().
+=== release 0.10.21 ===
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
+2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- NULL-terminate array of mpeg4 video file extensions.
- Fixes crash on PPC (#334226).
+ * 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
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
+2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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).
+ * 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
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- back to trunk
+ configure.ac: 0.10.20.4 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.20.4 pre-release
-=== release 0.10.4 ===
+2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- releasing 0.10.4, "Power"
+2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-10 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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.
+2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
-2006-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ 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
- Patch by: Fabrizio Gennari
+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).
- * 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)
+2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-03-10 Edward Hervey <edward@fluendo.com>
+ 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).
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix for variable not initialized.
+2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-03-09 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/libvisual/visual.c: (gst_visual_get_type),
- (gst_visual_src_setcaps), (gst_vis_src_negotiate),
- (gst_visual_chain):
- Small cleanups.
+2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2006-03-09 Wim Taymans <wim@fluendo.com>
+2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (audiocast_register_listener), (gst_gnome_vfs_src_start):
- Some cleanups.
+ 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).
-2006-03-09 Wim Taymans <wim@fluendo.com>
+2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
- Don't try to activate NULL chains.
+ 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.
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix invalid memory access to region before peek'd data (#332964).
+ 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.
-2006-03-09 Wim Taymans <wim@fluendo.com>
+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.
- Patch by: Christophe Fergeau
+2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
- Fix invalid memory access: make sure string passed to
- regexec() is NUL-termianted.
+ 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.
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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).
+ 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
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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).
+ 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.
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
- * 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/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.
-2006-03-08 Michael Smith <msmith@fluendo.com>
+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.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Make typefind function name for mpeg4 video unique.
+2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
-2006-03-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
- Use new basesink methods to configure max-lateness.
- Small doc update.
+ 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".
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
- Debug statement cleanups.
+2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- Simple cleanup.
+ 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.
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
- * 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.
+ 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.
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
+2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- Patch by: Fabrizio Gennari
+ 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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Allow palettes with less than 256 colours in AVI files
- (#333488)
+2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
-2006-03-07 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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/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).
+2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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().
+2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
- Don't ignore flow return from gst_pad_push().
+2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- Patch by: Fabrizio Gennari
+2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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)
+ 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.
-2006-03-06 Stefan Kost <ensonic@users.sf.net>
+2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- change location param details
- * gst/volume/gstvolume.c: (plugin_init):
- correct plugin description
+ 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.
-2006-03-06 Tim-Philipp Müller <tim at centricular dot net>
+2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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).
+ ext/vorbis/vorbisdec.c: Do not leak old taglist.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c:
+ Do not leak old taglist.
-2006-03-06 Wim Taymans <wim@fluendo.com>
+2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
- (gst_video_sink_class_init):
- Throw away frames that are later than 20 ms.
+ 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().
-2006-03-06 Tim-Philipp Müller <tim at centricular dot net>
+2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
- Patch by: Fabrizio Gennari
+ 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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Set depth on WMA caps (#333545)
+2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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.
+2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-05 Julien MOUTTE <julien@moutte.net>
+ 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'.
- * 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)
+2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for '3IVD' fourcc (#333403).
+2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-03-04 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * configure.ac:
- Bump requirements to GStreamer CVS for the new error enum.
+2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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).
+ gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Dist recently-added gstfastrandom.h.
-2006-03-04 Sebastien Moutte <sebastien@moutte.net>
+2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
- * win32/vs6:
- add a project file for libgstvolume
- update the workspace
+ 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 (?).
-2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ 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.
-2006-03-02 Wim Taymans <wim@fluendo.com>
+2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- Add audioresample to docs.
+ sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
+ Original commit message from CVS:
+ * sys/xvimage/Makefile.am:
+ floor() needs linking to $(LIBM).
- * gst/audioconvert/gstaudioconvert.c:
- Add revision date.
+2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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
-2006-03-02 Wim Taymans <wim@fluendo.com>
+2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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/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).
- * 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.
+2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2006-03-02 Wim Taymans <wim@fluendo.com>
+2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
- * 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.
+ 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.
- * 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.
+2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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.
-2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
+2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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.
-2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
+2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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/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).
-2006-03-02 Wim Taymans <wim@fluendo.com>
+2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
- * 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.
+ 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.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_type):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
- Fix docs for ffmpegcolorspace.
+2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-03-01 Tim-Philipp Müller <tim at centricular dot net>
+ 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/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).
+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.
-2006-03-01 Wim Taymans <wim@fluendo.com>
+2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * 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/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.
-2006-03-01 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/theora/Makefile.am:
- Dist new header too,
+2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
-2006-03-01 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/video/video.c: Revert ABI change.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: Revert ABI change.
- * 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.
+2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-03-01 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.
- * 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.
+2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ 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.
-2006-02-28 Sebastien Moutte <sebastien@moutte.net>
+2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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>
+ 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.
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- Fix some docs.
+2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-28 Edward Hervey <edward@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/alsa/gstalsasrc.c:
- Set proper class on the ElementDetails:
- Source/Audio instead of Src/Audio/
+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.
-2006-02-28 Edward Hervey <edward@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.
- * 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.
+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.
-2006-02-28 Wim Taymans <wim@fluendo.com>
+2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock):
- Don't try to provide a clock in the NULL state.
+ ChangeLog: ChangeLog surgery.
+ Original commit message from CVS:
+ * ChangeLog:
+ ChangeLog surgery.
+ * tests/examples/seek/seek.c:
+ Move variable into ifdef too.
-2006-02-28 Wim Taymans <wim@fluendo.com>
+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().
- * 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.
+2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-28 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- Don't ignore return code from ffmpeg convert function.
+2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
- Split out some long statements to ease debugging.
+ 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.
-2006-02-27 Jan Schmidt <thaytan@mad.scientist.com>
+2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
- Also, allow theoretical libvisual plugins that might support non-GL
- output even if they also do GL.
+2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-26 Julien MOUTTE <julien@moutte.net>
+ ChangeLog: Forgot to save the ChangeLog :/
+ Original commit message from CVS:
+ * ChangeLog:
+ Forgot to save the ChangeLog :/
- * 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.
+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.
-2006-02-25 Julien MOUTTE <julien@moutte.net>
+2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
+ 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.
-2006-02-24 Michael Smith <msmith@fluendo.com>
+2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * ChangeLog:
+ break long lines
+ Original commit message from CVS:
+ break long lines
- Add an MPEG4 video elementary stream typefind function.
+2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
-2006-02-04 Michael Smith <msmith@fluendo.com>
+ 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.
- * 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).
+2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- Patch by: Fabrizio Gennari <fabrizio dot get at tiscali dot it>
+2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Pick up palette for MS video v1 (#327028)
+ 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.
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
- * 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/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.
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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/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.
-2006-02-22 Wim Taymans <wim@fluendo.com>
+=== release 0.10.20 ===
- * ext/alsa/gstalsasrc.c:
- Fix description as pointed out by caugier.
+2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-02-22 Tommi Myöhänen <ext-tommi do myohanen at nokia dot 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
- Reviewed by : Edward Hervey <edward@fluendo.com>
+2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find):
- Better 3gp typefinding (#331526).
+ * 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:
+ * 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.
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
+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.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
- Don't send EOS event here, the base class will send one for us.
+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.
- * 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.
+2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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).
+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.
-2006-02-20 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)
- * 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).
+2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ 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...).
- * gst-libs/gst/audio/mixerutils.c:
- (element_factory_rank_compare_func):
- Make order in which elements are tried more determinable.
+2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-20 Julien MOUTTE <julien@moutte.net>
+ 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()
- * 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)
+2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-02-19 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
- use 0666 mask when creating files so umask gets applied
- correctly. Fixes #331295.
+2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-02-19 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/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).
+2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst/playback/gstdecodebin.c: (mimetype_is_raw):
- That should be text/x-pango-markup, not text/x-pango-layout.
+2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
-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.
+ 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.
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
- (gst_vorbisenc_convert_sink):
- Use some more scaling functions.
+ 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.
-2006-02-17 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@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/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.
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * 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.
-2006-02-17 Wim Taymans <wim@fluendo.com>
+2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst-libs/gst/audio/TODO:
- Update TODO
+ 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.
- * 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.
+2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-02-17 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/alsa/gstalsasink.c: (gst_alsasink_open),
- (gst_alsasink_reset):
- Also release lock when we get an error in _reset();
- fix an error message.
+2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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).
+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.
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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).
+ 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).
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/multichannel.c:
- Minor docs fix.
+ 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.
- * 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>
+2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
- Reviewed by Edward Hervey <edward@fluendo.com>
+ 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/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.
+2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-02-16 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-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).
+2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- Add gst-libs/gst/cdda to list of lib search paths.
+2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-15 Andy Wingo <wingo@pobox.com>
+ tests/check/libs/video.c: More checks.
+ Original commit message from CVS:
+ * tests/check/libs/video.c:
+ More checks.
- * 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.
+2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-15 Edward Hervey <edward@fluendo.com>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- quicktime movie files can also contain 'uuid' atoms.
+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.
-2006-02-14 Tim-Philipp Müller <tim at centricular dot net>
+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 ';'.
- * 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.
+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.
-2006-02-14 Wim Taymans <wim@fluendo.com>
+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.
- * 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
+2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-13 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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.
-2006-02-13 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+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.
-2006-02-13 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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>
+ 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.
- * 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.
+2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-12 Edward Hervey <edward@fluendo.com>
+ 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.
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Remove stray semi-colon (fixes #330888).
+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.
-2006-02-12 Jan Schmidt <thaytan@mad.scientist.com>
+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.
- * 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.
+2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
-2006-02-10 Edward Hervey <edward@fluendo.com>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Added flv file typefind (video/x-flv).
+2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-10 Edward Hervey <edward@fluendo.com>
+ 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.
- * 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.
+2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-02-09 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-09 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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!
+2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
+2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Back to CVS
+ 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).
-=== release 0.10.3 ===
+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 :)
-2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
+2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
- * configure.ac:
- releasing 0.10.3, "Under Pressure"
+ * 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.
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Drat. Bump libtool version number for new API.
- Prelease 0.10.2.3 (of 0.10.3)
+ 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.
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- * win32/common/config.h:
- 0.10.2.2 prerelease (of 0.10.3).
+ 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.
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
- Revert Andy's newsegment change pending a more correct
- fix.
+ 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.
-2006-02-08 Thomas Vander Stichele <thomas at apestaart dot org>
+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
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find), (plugin_init):
- detect more files as 3gp
- group and reorder the iso file formats
+2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * ext/vorbis/vorbis.c: (plugin_init):
- Register musicbrainz tags, so apps don't have to.
+2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
-2006-02-07 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-07 Tim-Philipp Müller <tim at centricular dot 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/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.
+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.
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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().
-2006-02-07 Andy Wingo <wingo@pobox.com>
+2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac (GST_MAJORMINOR): Update core version req to
- 0.10.2.2, for the collectpads API addition (#330244).
+ 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.
-2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- Return FALSE from plugin_init() when GnomeVFS can't
- be initialised for some reason (#328423).
+ 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
-2006-02-06 Julien MOUTTE <julien@moutte.net>
+2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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().
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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/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.
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
- (gst_vorbisenc_chain):
- Free some more leaked bits.
+2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * tests/check/pipelines/theoraenc.c: (start_pipeline),
- (stop_pipeline):
- Wait for state changes to happen if they're ASYNC.
+ gst/playback/gstdecodebin2.c: Fix signal docs.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Fix signal docs.
- This ought to teach those fancy pants buildbots a lesson.
+2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * gst-libs/gst/tag/gstid3tag.c:
- Add mapping for ID3 International Standard Recording Code
- tag "TSRC"
+2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
- Don't leak tag names.
+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>
-2006-02-05 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-02-05 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.
- * 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.
+2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+ 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.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
- (gst_text_overlay_init), (gst_text_overlay_src_event),
- (gst_text_overlay_collected): Fix clockoverlay.
+2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * docs/libs/compiling.sgml:
- Fix typo: it's pkg-config, not pkg-gconfig
+2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * 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.
+ 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.
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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?)
+ 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.
- * po/POTFILES.in:
- Add two translatable files.
+2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-05 Julien MOUTTE <julien@moutte.net>
+ ext/ogg/gstoggmux.c: Fix sample pipeline description.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Fix sample pipeline description.
- * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
- refactoring.
+2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-02-04 David Schleef <ds@schleef.org>
+ 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.
- * ext/ogg/gstoggmux.c:
- * gst/typefind/gsttypefindfunctions.c:
- Add Dirac typefinding and add dirac format to oggmux.
+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.
-2006-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2006-02-03 Edgard Lima <edgard.lima@indt.org.br>
+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.
- * 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.
+2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-02-03 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-02-02 Wim Taymans <wim@fluendo.com>
+ 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-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock):
- Ugh.. getting late I guess...
+2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-02-02 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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
-2006-02-02 Tim-Philipp Müller <tim at centricular dot net>
+=== release 0.10.18 ===
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Unlinking two source pads is ... hard.
+2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2006-02-02 Wim Taymans <wim@fluendo.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/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
- * gst-libs/gst/audio/TODO:
- Updated.
+2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ * 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.
-2006-02-01 Philippe Kalaf <burger at speedy dot org>
+2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- 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.
+ 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.
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
- * 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.
+ 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.
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
- * 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/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.
-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.
+2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
-
+2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
-2006-01-31 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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.
-2006-01-31 Wim Taymans <wim@fluendo.com>
+2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2006-01-31 Wim Taymans <wim@fluendo.com>
+2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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)
-2006-01-31 Stefan Kost <ensonic@users.sf.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/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_create_sine_table), (plugin_init):
- * gst/volume/gstvolume.c: (plugin_init):
- initialize gst_controller before using
+2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-31 Jan Schmidt <thaytan@mad.scientist.com>
+ 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).
- * 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.
+2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- Fixes failing tests.
+ 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.
-2006-01-31 Andy Wingo <wingo@pobox.com>
+2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
- * 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.
+2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
- fraction, not double.
+ 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.
-2006-01-31 Sebastien Moutte <sebastien@moutte.net>
+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.
- * win32/vs7:
- add vs7 project files created by Sergey Scobich
+2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-30 Sebastien Moutte <sebastien@moutte.net>
+ 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).
- * win32/vs8:
- add vs8 project files created by Sergey Scobich
-
-2006-01-30 Andy Wingo <wingo@pobox.com>
+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/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.
+ * 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().
- * 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 +.
+2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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.
-2006-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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/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.
-2006-01-30 Andy Wingo <wingo@pobox.com>
+2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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/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.
- * 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.
+2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
- * tests/check/Makefile.am:
- * tests/check/pipelines/theoraenc.c: New file, basically does same
- tests as vorbisenc.
+ 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).
- * tests/check/pipelines/vorbisenc.c: I claim these bugs.
+2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-01-30 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
- (gst_audioringbuffer_pause):
- Implement pause that does not wait for completion.
+2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
- * 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>
+ 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.
- * 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)
+2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-01-30 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-01-29 Sebastien Moutte <sebastien@moutte.net>
+ 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.
- * 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>
+2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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.
-2006-01-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * win32/MANIFEST:
- * win32/common/config.h:
- * win32/common/config.h.in:
- add config files just like in core
+ 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'.
-2006-01-28 Tim-Philipp Müller <tim at centricular dot net>
+2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
- Make gcc-4.1 happy (part of #327357).
+ 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.
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- check for and throw RESOURCE_BUSY
+ 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.
-2006-01-27 David Schleef <ds@schleef.org>
+2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2006-01-27 David Schleef <ds@schleef.org>
+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.
- * REQUIREMENTS:
- * configure.ac: update liboil requirement to 0.3.6
- * gst/videoscale/Makefile.am:
- * gst/videoscale/vs_scanline.c: liboilify
+2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * ext/libvisual/visual.c: (get_buffer):
- When pad_alloc returns a GstFlowReturn other
- than GST_FLOW_OK, make sure it is passed upstream.
-
-2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
+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.
-2006-01-27 Thomas Vander Stichele <thomas at apestaart dot org>
+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/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
+2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-27 Julien MOUTTE <julien@moutte.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/subparse/gstsubparse.c: (gst_subparse_type_find),
- (plugin_init): Make typefinding of subtitles work again.
+2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-26 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
-2006-01-26 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-26 Edward Hervey <edward@fluendo.com>
+ 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).
- * gst/playback/gstdecodebin.c: (pad_probe):
- Also consider the flush-start and tag events as unblockers
- for the pad probes.
+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.
-2006-01-26 Julien MOUTTE <julien@moutte.net>
+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.
- * 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.
+2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
-2006-01-25 Wim Taymans <wim@fluendo.com>
+ * 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
- * 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.
+2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-01-25 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
-2006-01-23 Edward Hervey <edward@fluendo.com>
+ 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.
- * 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
+2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-01-23 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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>
+2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
-2006-01-22 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
-2006-01-22 Tim-Philipp Müller <tim at centricular dot 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.
- * 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>
+2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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).
-2006-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Free iterator when done.
+ 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.
-2006-01-17 Jan Schmidt <thaytan@mad.scientist.com>
+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.
- * 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.
+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.
- Makes this work again:
+2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- 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
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
-=== release 0.10.2 ===
+=== release 0.10.17 ===
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 0.10.2, "Then the devil is six"
+ * 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.
-2006-01-16 Jan Schmidt <thaytan@mad.scientist.com>
+2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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.
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix #327216 (null dereference in vorbisdec)
+ 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.
-2006-01-16 Tim-Philipp Müller <tim at centricular dot net>
+2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/theora/theoradec.c: (theora_handle_comment_packet):
- Post taglist actually on bus instead of just freeing it
- (fixes #327114 and totem bug #327080).
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * 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.
+=== release 0.10.16 ===
-2006-01-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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
+ * 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
-2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
- fix a silly segfault
+ * 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.
-2006-01-14 Tim-Philipp Müller <tim at centricular dot net>
+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
- * 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.
+2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-13 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix playback for sources that emit raw audio or
- raw video streams (e.g.: cd audio sources) (#325984).
+2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * gst-libs/gst/audio/mixerutils.c:
- (gst_audio_mixer_filter_do_filter):
- actually save the element we create
+2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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.
+2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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.
+2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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).
+2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected):
- Make debug output more useful by using GST_PTR_FORMAT.
+ 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).
-2006-01-11 Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
- Set depth and width for alaw/mulaw (fixes #326601).
+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.
-2006-01-11 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * tests/icles/Makefile.am:
- don't build the tests if we don't have the libs
+2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+ docs/: Add *-undeclared.txt to fix buildbot.
+ Original commit message from CVS:
+ * docs/libs/.cvsignore:
+ * docs/plugins/.cvsignore:
+ Add *-undeclared.txt to fix buildbot.
- * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
- (gst_cd_paranoia_paranoia_callback):
- Don't try to free NULL pointers.
+2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
-2006-01-10 Edward Hervey <edward@fluendo.com>
+ 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.
- * 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.
+2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
-2006-01-10 Michael Smith <msmith@fluendo.com>
+ 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.
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
- Don't leak GCond in audio sources.
-
-2006-01-10 Jan Schmidt <thaytan@mad.scientist.com>
+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).
- * gst/playback/gstplaybin.c: (gen_audio_element):
- Don't leak an autoaudiosink/alsasink when we generate
- a new audio element. (old code, I guess)
+2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
-2006-01-10 Michael Smith <msmith@fluendo.com>
+ 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.
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
- Support float audio in audiorate.
- Use width rather than depth for selecting sample width.
+2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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).
+2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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
- * gst-libs/gst/audio/Makefile.am:
- Link against libgstinterfaces, needed for mixer
- and property probe stuff.
+2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-09 Edward Hervey <edward@fluendo.com>
+ gst/playback/.cvsignore: Ignore more.
+ Original commit message from CVS:
+ * gst/playback/.cvsignore:
+ Ignore more.
- * gst-libs/gst/Makefile.am:
+2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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-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.
+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.
-2006-01-03 Michael Smith <msmith@fluendo.com>
+2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/audioresample/resample.h:
- As before, but for o_buf
+ 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.
-2006-01-03 Michael Smith <msmith@fluendo.com>
+2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
- * gst/audioresample/resample.h:
- Declare struct _ResampleState.buffer as unsigned char *, not void *,
- since we do arithmetic on it.
+ 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.
-2006-01-02 Tim-Philipp Müller <tim at centricular dot net>
+2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
- * 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).
+ 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.
-2005-12-30 Stefan Kost <ensonic@users.sf.net>
+2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ Makefile.am: Use new common/win32.mak.
+ Original commit message from CVS:
+ * Makefile.am:
+ Use new common/win32.mak.
-2005-12-29 Stefan Kost <ensonic@users.sf.net>
+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/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
+2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot 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.
- * po/POTFILES.in:
- ... and this time the other modified file that I missed last time.
+2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-29 Michael Smith <msmith@fluendo.com>
+ 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).
- * gst/playback/gstdecodebin.c: (new_pad):
- Fix non-C89 variable declaration not at the start of a block. Should
- help some compilers.
+2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * tests/check/Makefile.am:
- And now fix 'make distcheck' (builddir != srcdir)
+2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * tests/check/Makefile.am:
- Fix test case compilation.
+2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
- * tests/check/Makefile.am:
- Add cddabasesrc unit test; also actually enable the vorbis test.
+ 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.
- * tests/check/generic/states.c:
- Blacklist new cd audio elements as well.
+2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
- * tests/check/libs/cddabasesrc.c:
- Unit test for GstCddaBaseSrc (discid calculation mostly).
+ 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.
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/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.
- * 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.
+2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+ 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?).
- * 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.
+2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * ext/gnomevfs/gstgnomevfssink.h:
- Use GstBaseSinkClass as parent_class member for class struct, not
- GstBaseSink.
+2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+ * ChangeLog:
+ ChangeLog: remove conflict markers
+ Original commit message from CVS:
+ ChangeLog: remove conflict markers
- Patch by: Michael Benes
+2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.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)
+ 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.
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/audioconvert/Makefile.am: Also link to libm.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Also link to libm.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix example pipeline in gtk-doc blurb.
+2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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.
-=== release 0.10.1 ===
+2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- releasing 0.10.1, "Dobro Dedek"
+2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
-2005-12-21 Edgard Lima <edgard.lima@indt.org.br>
+ 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/typefind/gsttypefindfunctions.c:
- iLBC30 and iLBC20 added to typefind.
+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.
-2005-12-20 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * 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
+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).
-2005-12-20 Edward Hervey <edward@fluendo.com>
+2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ gst-plugins-base.doap: Oops, fix the release name.
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Oops, fix the release name.
-2005-12-19 Edward Hervey <edward@fluendo.com>
+2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.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-plugins-base.doap: Add 0.10.15 release
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Add 0.10.15 release
-2005-12-18 Jan Schmidt <thaytan@mad.scientist.com>
+2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.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: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * 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
+=== release 0.10.15 ===
-2005-12-18 Julien MOUTTE <julien@moutte.net>
+2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
- (gst_video_sink_get_type): Add a debug category.
+ 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"
-2005-12-17 Philippe Khalaf <burger@speedy.org>
+2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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)
+ * 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.
-2005-12-17 Philippe Khalaf <burger@speedy.org>
+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).
- * 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)
+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).
-2005-12-16 Stefan Kost <ensonic@users.sf.net>
+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).
- * 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
+2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * tests/check/generic/states.c: (GST_START_TEST):
- Blacklist cdparanoia element in state test.
+2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- Patch by: Benjamin Pineau
+2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
- * 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)
+ 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.
-2005-12-15 Michael Smith <msmith@fluendo.com>
+2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
- * 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-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).
-2005-12-15 Michael Smith <msmith@fluendo.com>
+2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
- * gst/audioresample/gstaudioresample.c:
- Don't leak all input buffers to audioresample.
+ 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).
-2005-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.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-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.
-2005-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.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.
-
-2005-12-14 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * 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
+2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-14 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsamixer.h:
- Comment out GST_ALSA_MIXER, it returns a struct that's not
- used.
+ 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.
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Add some debug info.
+2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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).
+ 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.
-2005-12-12 Christian Schaller <uraeus@gnome.org>
+2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/Makefile.am: fix cdparanoia entry
- * gst-plugins-base.spec.in: add cdparanoia
+ 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.
-2005-12-12 Michael Smith <msmith@fluendo.com>
+2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Use the correct function to free list of typefind factories.
+ 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.
-2005-12-12 Wim Taymans <wim@fluendo.com>
+2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ tests/icles/stress-playbin.c: Fix the build.
+ Original commit message from CVS:
+ * tests/icles/stress-playbin.c:
+ Fix the build.
-2005-12-12 Wim Taymans <wim@fluendo.com>
+2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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..
+ 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.
-2005-12-12 Wim Taymans <wim@fluendo.com>
+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>
- * tests/examples/seek/scrubby.c: (main):
- Set higher priority for bus events so they don't get reordered with
- gtk gui events.
+ * 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.
- * 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.
+2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
-2005-12-12 Tim-Philipp Müller <tim at centricular dot net>
+ * 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).
- * 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.
+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).
- * tools/.cvsignore:
- Ignore more.
+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.
-2005-12-11 Julien MOUTTE <julien@moutte.net>
+2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-12-09 Michael Smith <msmith@fluendo.com>
+2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * tools/Makefile.am:
- * tools/gst-launch-ext-m.m:
- Remove gst-launch-ext. It doesn't work, and is no longer
- particularly useful.
+ 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.
-2005-12-08 Luca Ognibene <luogni@tin.it>
+2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ * ChangeLog:
+ Remove stray character from the changelog.
+ Original commit message from CVS:
+ Remove stray character from the changelog.
-2005-12-07 Tim-Philipp Müller <tim at centricular dot net>
+2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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>
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
- 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.
+2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-06 Thomas Vander Stichele <thomas at apestaart dot 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.
- * 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:
- * 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
+2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
-=== release 0.10.0 ===
+ 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).
-2005-12-05 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- releasing 0.10.0, "Mont-d'or"
+ 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.
-2005-12-05 Jan Schmidt <thaytan@mad.scientist.com>
+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.
- * tests/examples/seek/Makefile.am:
- Build fix for when gtk is not available.
+2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-12-05 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-12-05 Andy Wingo <wingo@pobox.com>
+ 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.
- patch by: Kai Vehmanen <kv2004 eca cx>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c
- (gst_base_rtp_depayload_thread): Fix busy loop (#323017).
+2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-12-04 Andy Wingo <wingo@pobox.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.
- patch by: Sebastien Cote <sebas642 yahoo ca>
-
- * pkgconfig/gstreamer-plugins-base.pc.in (Libs): Add -L flag.
- Fixes #319172.
+2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-12-02 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-12-02 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-12-02 Wim Taymans <wim@fluendo.com>
+ tests/examples/snapshot/snapshot.c: Fix the build.
+ Original commit message from CVS:
+ * tests/examples/snapshot/snapshot.c:
+ Fix the build.
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Small cleanups.
+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.
-2005-12-02 Wim Taymans <wim@fluendo.com>
+2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- Post errors.
+ 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.
-=== release 0.9.7 ===
+2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ 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.
- * configure.ac:
- releasing 0.9.7, "Mi Perro No Tiene Ninguna Nariz"
+2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * Makefile.am:
- * po/hu.po:
- * win32/MANIFEST:
- * win32/gst.sln:
- add win32 MANIFEST file
- do something to the hungarian translation
+2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-12-01 Tim-Philipp Müller <tim at centricular dot net>
+ * po/LINGUAS:
+ add languages
+ Original commit message from CVS:
+ add languages
- * ext/Makefile.am:
- Add $(PANGO_DIR) to SUBDIRS
+2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttimeoverlay.c:
- Fix and improve element descriptions.
+ 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.
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ po/fi.po: Added Finnish translation.
+ Original commit message from CVS:
+ translated by: Ilkka Tuohela <hile@iki.fi>
+ * po/fi.po:
+ Added Finnish translation.
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- * ext/Makefile.am:
- moved pango to base
+ 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.
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ po/da.po: Added Danish translation.
+ Original commit message from CVS:
+ translated by: Mogens Jaeger <mogens@jaeger.tf>
+ * po/da.po:
+ Added Danish translation.
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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.
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * win32/:
- add Visual Studio 6 build files
+ po/bg.po: Added Bulgarian translation.
+ Original commit message from CVS:
+ translated by: Alexander Shopov <ash@contact.bg>
+ * po/bg.po:
+ Added Bulgarian translation.
-2005-12-01 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.
- * 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>
+2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins.args:
- * 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
-
-2005-11-30 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2005-11-30 Julien MOUTTE <julien@moutte.net>
+ Ignore registries in any format.
+ Original commit message from CVS:
+ * docs/plugins/.cvsignore:
+ * tests/check/.cvsignore:
+ Ignore registries in any format.
- * 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.
+2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ 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.
- * LICENSE:
- move
- * 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:
- * Makefile.am:
- update
- * scripts/autoplugins.sh:
- remove
+2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ 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.
- * 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/
+2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-30 Edward Hervey <edward@fluendo.com>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- Go away you stupid GstStaticPadTemplate memleak.
+2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot 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-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"
+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.
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
+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).
- * 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
+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.
-2005-11-30 Edward Hervey <edward@fluendo.com>
+2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
- * gst-libs/gst/netbuffer/Makefile.am:
- really, really tired
+ 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.
-2005-11-30 Edward Hervey <edward@fluendo.com>
+2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Update for new GstTypeFindFactory _register()
+ 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.
-2005-11-30 Edward Hervey <edward@fluendo.com>
+2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
- * 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)
+ 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).
-2005-11-29 Andy Wingo <wingo@pobox.com>
+2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
- * pkgconfig/gstreamer-plugins-base.pc.in:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
- (#322257).
+ * 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.
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gststreamselector.c: (gst_stream_selector_chain):
- 3rd time's the charm. Correct ref-counting for discarded buffers.
+ configure.ac: Fix configure check for HAVE_LIBXML_HTML.
+ Original commit message from CVS:
+ * configure.ac:
+ Fix configure check for HAVE_LIBXML_HTML.
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2005-11-29 Tim-Philipp Müller <tim at centricular dot net>
+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.
- * gst/subparse/gstsubparse.c: (feed_textbuf):
- Don't access already unref'ed buffer.
+2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+ gst/playback/gststreaminfo.c: Fix build.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c:
+ Fix build.
- * 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
+2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-29 Julien MOUTTE <julien@moutte.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find): Free the list,
- patch from bug #322704 (Alessandro Decina).
+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.
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
- * gst-libs/gst/audio/Makefile.am:
- folded audiofilter into the audio library
+ 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.
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/gstvideoscale.c:
- remove unimplemented scale methods
+ 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.
-2005-11-28 Tim-Philipp Müller <tim at centricular dot net>
+2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
- Don't leak caps.
+ 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).
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.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 !
+ 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).
-2005-11-28 Edward Hervey <edward@fluendo.com>
+2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-11-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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
+ 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.
-2005-11-28 Wim Taymans <wim@fluendo.com>
+2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (group_commit), (probe_triggered):
- * gst/playback/gstplaybasebin.h:
- Prepare to handle errors betters.
+ 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.
- * 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.
+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.
-2005-11-28 Wim Taymans <wim@fluendo.com>
+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
- * gst-libs/gst/audio/TODO:
- Updated TODO
+2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_open_device),
- (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release):
- Small cleanups.
+ * ChangeLog:
+ * 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:
+ 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-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.
+2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Fix navigation events
- coordinates translation with pixel aspect ratios.
+ 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.
-2005-11-28 Julien MOUTTE <julien@moutte.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.
- * 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)
+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.
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
- * 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.
+ 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.
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Refuse to create an XvImage if we can't find the format.
+ * 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)
-2005-11-28 Edward Hervey <edward@fluendo.com>
+2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.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.
+ 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).
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds):
- Don't try to create a zero-sized subbuffer.
+ 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.
-2005-11-27 Julien MOUTTE <julien@moutte.net>
+2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/: Printf format fixes (#465028).
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c:
+ * gst/videorate/gstvideorate.c:
+ Printf format fixes (#465028).
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.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/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.
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.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
+ 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.
-2005-11-27 Julien MOUTTE <julien@moutte.net>
+2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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.
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
- * 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.
+ 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
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+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
- * 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.
+=== release 0.10.14 ===
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * 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-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.
+ * 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
-2005-11-26 Edward Hervey <edward@fluendo.com>
+2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_chain):
- Properly return GstFlowReturn from gst_pad_push in chain functions.
+ * 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
+
+2007-07-27 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.
-2005-11-25 Michael Smith <msmith@fluendo.com>
+2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.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.
+ 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.
-2005-11-24 Andy Wingo <wingo@pobox.com>
+2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_handle_client_write):
- Be threadsafe.
+ 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.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * configure.ac: back to HEAD
+2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-=== release 0.9.6 ===
+ tests/: We should use $(LIBM).
+ Original commit message from CVS:
+ * tests/examples/volume/Makefile.am:
+ * tests/icles/Makefile.am:
+ We should use $(LIBM).
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- releasing 0.9.6, "White Eight"
+ tests/icles/Makefile.am: This needs -lm.
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ This needs -lm.
-2005-11-23 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.
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- * docs/plugins/inspect/plugin-sine.xml:
- remove sinesrc some more
+2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
- If we are reading too slowly, jump forward in the ringbuffer
- instead of blocking.
+2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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
+2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-23 Michael Smith <msmith@fluendo.com>
+ 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.
- * 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_*().
+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.
-2005-11-23 Edward Hervey <edward@fluendo.com>
+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.
- * autogen.sh:
- eheh, sinesrc is gone. Replacing sample srcfile by existing one.
- Fixes autogen
+2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * check/Makefile.am:
- * check/elements/videotestsrc.c: (setup_videotestsrc),
- (cleanup_videotestsrc), (GST_START_TEST), (videotestsrc_suite),
- (main):
- add a test for videotestsrc
+2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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:
- * gst-plugins-base.spec.in:
- remove sinesrc from the build
+2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+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.
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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:
- Translation string updates
+ 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).
- * 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.
+2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * ext/libvisual/visual.c: (gst_visual_chain):
- Fix the fps calculations.
+2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/ffmpegcolorspace/avcodec.h:
- Move structure element for clarity
+ MAINTAINERS: Updating all the maintainers files
+ Original commit message from CVS:
+ * MAINTAINERS:
+ Updating all the maintainers files
- * 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...
+2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-22 Wim Taymans <wim@fluendo.com>
+ 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-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...
+2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
-2005-11-22 Wim Taymans <wim@fluendo.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>
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
- This clock can be slaved to a master clock now.
+2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
+ * ChangeLog:
* 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.
-
- * 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.
-
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2005-11-22 Michael Smith <msmith@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- Add underscore.
+ * 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.
-2005-11-22 Michael Smith <msmith@fluendo.com>
+2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
- * 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/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.
-2005-11-22 Jan Schmidt <thaytan@mad.scientist.com>
+2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ po/vi.po: Update translations.
+ Original commit message from CVS:
+ * po/vi.po:
+ Update translations.
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- remove some deprecated functions
+ 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.
-2005-11-22 Andy Wingo <wingo@pobox.com>
+2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
- * Update for gst_tag_setter API changes.
+ 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.
-2005-11-22 Andy Wingo <wingo@pobox.com>
+2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-11-22 Wim Taymans <wim@fluendo.com>
+2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * examples/seeking/seek.c: (main):
- Give higher priority to bus signals than the gtk events
- to fix a race condition in the segment looping.
+ 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.
-2005-11-22 Tim-Philipp Müller <tim at centricular dot 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/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-plugins-base.spec.in:
- Rename libgsttagedit to libgsttag (#322117).
+2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-22 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+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.
-2005-11-22 Julien MOUTTE <julien@moutte.net>
+2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Handle navigation
- events correcly with borders if applicable.
+ 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.
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- Patch by: Luca Ognibene
+ 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.
- * 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
+2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
-2005-11-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+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
- * sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
- (gst_xvimagesink_xoverlay_init):
- Remove desired size stuff (#321816).
+=== release 0.10.13 ===
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ * 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?"
-2005-11-21 Andy Wingo <wingo@pobox.com>
+2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.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.
+ * 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.
-2005-11-21 Andy Wingo <wingo@pobox.com>
+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/playback/gstplaybin.c (gen_audio_element)
- (gen_video_element): Use the new MISSING_PLUGIN core error
- category. Closes #320060.
+2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
- * 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.
+ 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.
- * gst/subparse/gstsubparse.c (gst_subparse_src_event):
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek): Update for
- stream lock changes.
+2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-21 Wim Taymans <wim@fluendo.com>
+ ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ Repair umlaut.
- * 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.
+2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-11-21 Andy Wingo <wingo@pobox.com>
+ 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.
- * *.h:
- * *.c: Ran scripts/update-macros. Oh yes.
+2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
-2005-11-21 Michael Smith <msmith@fluendo.com>
+ 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)
- * 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.
+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>
-2005-11-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-plugins-base.spec.in:
+ update spec
+ Original commit message from CVS:
+ update spec
- * 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)
+2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+ 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.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
- * 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.
+ * 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).
-2005-11-21 Andy Wingo <wingo@pobox.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.
- patch by: Alessandro Dessina <alessandro nnva org>
+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.
- * 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.
+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.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
- * 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.
+ 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.
-2005-11-21 Michael Smith <msmith@fluendo.com>
+2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
- (gst_vorbisenc_chain), (gst_vorbisenc_output_buffers):
- Properly handle pad_push return values.
+ 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.
-2005-11-20 Jan Schmidt <thaytan@mad.scientist.com>
+2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@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>
+ 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-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Fix the audiosrc base class again, we did not unflush.
+2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
-2005-11-17 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-17 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ 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
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_configure_segment),
- (gst_ogg_demux_perform_seek):
- Segment done must include stream time.
+2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
- * 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/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.
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init):
- Fix compile again.
+ 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.
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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.
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
- * check/generic/states.c: (GST_START_TEST):
- fix the test so that it only checks for elements that are part of
- this source module
+ 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.
-2005-11-16 Michael Smith <msmith@fluendo.com>
+2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ 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.
-2005-11-16 Edward Hervey <edward@fluendo.com>
+2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_event), (gst_videorate_chain):
- Handle segment seeks
+ 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!
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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-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.
+2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-16 Julien MOUTTE <julien@moutte.net>
+ * 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-libs/gst/video/gstvideosink.c: (gst_video_sink_center_rect):
- * gst-libs/gst/video/videosink.h: Add helper function needed
- for video sinks.
+2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
-2005-11-16 Tim-Philipp Müller <tim at centricular dot net>
+ 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/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- Don't leak reference to pad parent.
+2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-16 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.
- * examples/seeking/seek.c: (send_event), (do_seek),
- (loop_toggle_cb), (segment_done), (main):
- Added looping checkbox.
+2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-15 Johan Dahlin <johan@gnome.org>
+ 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.
- * 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.
+2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- Make state change function thread safe.
-
-2005-11-15 Edward Hervey <edward@fluendo.com>
-
- * 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
-
-2005-11-14 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2005-11-14 Michael Smith <msmith@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_client_queue_data),
- (gst_multifdsink_render):
- Don't leak GDP headers when using GDP mode (i.e. tcpserversink).
-
-2005-11-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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>
-
- * gst/playback/gstplaybin.c (gen_audio_element): Use autoaudiosink
- as well if it is available. Fixes #316442.
-
-2005-11-14 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2005-11-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2005-11-13 Edward Hervey <edward@fluendo.com>
+ 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).
- * 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.
+2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac: back to HEAD
+2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
-=== release 0.9.5 ===
+ 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.
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- releasing 0.9.5, "No No Kia"
+ 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.
-2005-11-11 Edward Hervey <edward@fluendo.com>
+2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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"
+ 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.
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+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).
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Check whether peer pad exists before sending navigation events
- to it.
+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).
-2005-11-11 Michael Smith <msmith@fluendo.com>
+2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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.
+ 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.
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
- * 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.
+ 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.
-2005-11-10 Andy Wingo <wingo@pobox.com>
+2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstplaybasebin.c (prepare_output): Fix format
- string doober.
+ 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
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps):
- Prefer passthrough in transform_caps
+ * 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).
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-11-10 Wim Taymans <wim@fluendo.com>
+2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
- * 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.
+ 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).
-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.
+2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-10 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * 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
+2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-10 Tim-Philipp Müller <tim at centricular dot 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/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.
+2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
-2005-11-10 Michael Smith <msmith@fluendo.com>
+ 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).
- * 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.
+2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-09 Tim-Philipp Müller <tim at centricular dot net>
+ tests/check/pipelines/.cvsignore: Fix build.
+ Original commit message from CVS:
+ * tests/check/pipelines/.cvsignore:
+ Fix build.
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose):
- Set elements to NULL state before disposing of them.
+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>
-2005-11-08 Sebastien Cote <sebas642 at yahoo dot ca>
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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>
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
- * gst/audioconvert/gstaudioconvert.c:
- Fix typo in docs.
+2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-11-08 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-11-08 Brian Cameron <brian dot cameron at sun dot com>
+ 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.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/volume/gstvolume.c: (volume_set_caps):
- Fix compilation on Solaris with Forte. (#320923)
+ * gst/tcp/gstmultifdsink.c:
+ add debug
+ Original commit message from CVS:
+ add debug
-2005-11-08 Wim Taymans <wim@fluendo.com>
+2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- No need to do a typecheck.
+ * 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
-2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
+2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.h:
- We register a debug category, so let's use it.
+ * ext/ogg/gstoggmux.c:
+ * gst/gdp/gstgdpdepay.c:
+ debug changes
+ Original commit message from CVS:
+ debug changes
-2005-11-04 Zeeshan Ali <zeenix@gmail.com>
+2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_wait):
- Fixed a small problem.
+ 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.
-2005-11-04 Wim Taymans <wim@fluendo.com>
+2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ ext/theora/theoradec.c: Fix misleading comment.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_sink_event):
+ Fix misleading comment.
- * 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.
+2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-11-04 Zeeshan Ali <zeenix@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.
- * 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.
+2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-11-03 Wim Taymans <wim@fluendo.com>
+ 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/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.
+2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-11-02 Sebastien Cote <sebas642@yahoo.ca>
+ 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.
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.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):
- 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>
+ 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.
- * 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.
+2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+ 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.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push):
- Fixed the queueing algorithm.
+2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+ * gst/tcp/gstmultifdsink.c:
+ log tweaking
+ Original commit message from CVS:
+ log tweaking
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_push):
- A small fix
+2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-10-31 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.
- * gst-libs/gst/audio/gstringbuffer.h:
- Don't break ABI.
+2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * 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
-2005-10-31 Wim Taymans <wim@fluendo.com>
+2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ adding debugging
+ Original commit message from CVS:
+ adding debugging
-2005-10-29 Tim-Philipp Müller <tim at centricular dot net>
+2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
- * 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)
+ * common:
+ * gst-plugins-base.spec.in:
+ update spec file for RTP changes
+ Original commit message from CVS:
+ update spec file for RTP changes
-2005-10-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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.
-2005-10-27 Philippe Khalaf <burger at speedy dot org>
+2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/rtp/gstbasedepayload.c:
- * gst-libs/gst/rtp/gstbasedepayload.h:
- Minor cleanups
+ 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.
-2005-10-27 Tim-Philipp Müller <tim at centricular dot net>
+2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/.cvsignore:
- * gst/playback/decodetest.c:
- * gst/playback/test3.c:
- Port these two tests as well.
+ 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.
-2005-10-27 Wim Taymans <wim@fluendo.com>
+2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * 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.
+ 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.
-2005-10-27 Wim Taymans <wim@fluendo.com>
+2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
- * gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
- (gst_adder_change_state):
- Fix timestamps and fix deadlock when stopping the collectpads.
+ 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.
-2005-10-27 Zeeshan Ali <zeenix@gmail.com>
+2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Declaring the payload types as strings too so that they can be used
- in the padtemplate inialization.
+ 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.
-2005-10-26 Andre Magalhaes <andrunko@gmail.com>
+2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+ 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.
- * 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.
+2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-26 Michael Smith <msmith@fluendo.com>
+ 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/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.
+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.
-2005-10-26 Zeeshan Ali <zeenix@gmail.com>
+2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-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
+2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-26 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/pipelines/vorbisdec.c:
+ commit new file
+ Original commit message from CVS:
+ commit new file
- * 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.
+2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
-2005-10-25 Zeeshan Ali <zeenix@gmail.com>
+ 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).
- * 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.
+2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-24 Julien MOUTTE <julien@moutte.net>
+ 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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Check
- the caps against our xcontext caps.
+2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.spec.in:
+ update spec file
+ Original commit message from CVS:
+ update spec file
- * 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.
+2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
-2005-10-24 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-24 Julien MOUTTE <julien@moutte.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.
- * 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.
+2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- back to HEAD
+2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
-=== release 0.9.4 ===
+ 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.
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.9.4, "Velociraptor"
+ 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.
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- * po/POTFILES.in:
- STOPPED -> FAILED
+ 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.
-2005-10-21 Wim Taymans <wim@fluendo.com>
+2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-10-21 Wim Taymans <wim@fluendo.com>
+2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/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.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Don't try to sync on buffers without a timestamp.
+2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
-2005-10-21 Wim Taymans <wim@fluendo.com>
+ 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/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.
+2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
-2005-10-20 Tim-Philipp Müller <tim at centricular dot 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/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types),
- (gst_vorbisenc_src_query):
- Implement position and duration queries.
+2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/test3.c: (update_scale), (main):
- Fix for async state changes and print nicer output.
+ 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.
-2005-10-20 Wim Taymans <wim@fluendo.com>
+2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- * gst/playback/test6.c: (main):
- Fix tests again
+ 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.
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+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.
- * 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.
+2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/gdp/gstgdpdepay.c:
+ add buffer logging
+ Original commit message from CVS:
+ add buffer logging
- * 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.
+2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-20 Wim Taymans <wim@fluendo.com>
+ 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
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gst_play_base_bin_change_state):
- Fix leak.
- Handle case where playbasebin is now ASYNC because
- decodebin is.
+2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioresample/gstaudioresample.c:
+ add debugging and reformat docs
+ Original commit message from CVS:
+ add debugging and reformat docs
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/bufferframesconvert.c:
- * gst/audioconvert/plugin.c: (plugin_init):
- * gst/audioconvert/plugin.h:
- And bye bye buffer-frames-convert
+2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
-2005-10-19 Wim Taymans <wim@fluendo.com>
+ 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...
- * 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>
+2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
- * 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.
+ * 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.
- * 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.
+2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-17 Stefan Kost <ensonic@users.sf.net>
+ 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).
- * gst/audiotestsrc/Makefile.am:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- fix broken build of controllerized plugins
+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).
-2005-10-17 Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot 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.
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for Indeo-3 (IV32).
+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.
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- rewrite
+ 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.
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/video/video.c: (gst_video_get_size):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- doc updates
+ 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
-2005-10-17 Andy Wingo <wingo@pobox.com>
+2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
- * gst/tcp/gstmultifdsink.c: Convert to use the boilerplate macro.
+2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcp.c (gst_tcp_socket_read): Comment update.
+ 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.
- * 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.
+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/ogg/gstoggmux.c (gst_ogg_mux_send_headers): Fix a segfault
- in strange circumstance.
+2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-10-17 Julien MOUTTE <julien@moutte.net>
+ 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.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size): We are asked to compute a buffer size
- from caps, let's use the caps...
+2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-17 Thomas Vander Stichele <thomas at apestaart 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
- * configure.ac:
- put back AX_CREATE_STDINT_H, ffmpegcolorspace includes _stdint.h
+2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/tcp/gstmultifdsink.c:
+ update docs
+ Original commit message from CVS:
+ update docs
- * gst-libs/gst/interfaces/Makefile.am:
- fix silly typo
+2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-16 Andy Wingo <wingo@pobox.com>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * gst/playback/gstdecodebin.c
- (gst_element_set_state_like_a_crazy_man): New kraaaaaaazy
- function!
- (try_to_link_1): Increase kraziness level.
+=== release 0.10.12 ===
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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
-
-2005-10-13 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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.
+2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-13 Stefan Kost <ensonic@users.sf.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
- * examples/stats/mp2ogg.c:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- typo fixes
+2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-13 Michael Smith <msmith@fluendo.com>
+ 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
- * ext/ogg/gstoggmux.c:
- Use magic glib macros to define constants as 64 bit, to ensure
- appropriate vararg passing.
+2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-10-13 Michael Smith <msmith@fluendo.com>
+ 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.
- * 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.
+2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-12 Stefan Kost <ensonic@users.sf.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
- * 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
+2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-10-12 Wim Taymans <wim@fluendo.com>
+ 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.
- * examples/seeking/Makefile.am:
- Oops.
+2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-12 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-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read), (gst_ring_buffer_clear):
- Don't assert on normal stuff.
+2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybin.c: (do_playbin_seek):
- API fix.
+ * tests/check/libs/.gitignore:
+ moap ignore
+ Original commit message from CVS:
+ moap ignore
-2005-10-12 Wim Taymans <wim@fluendo.com>
+2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * win32/common/config.h:
+ * win32/common/libgstutils.def:
+ update defs
+ Original commit message from CVS:
+ update defs
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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
-2005-10-11 Wim Taymans <wim@fluendo.com>
+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.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Respect segment rate and accum when scheduling samples.
+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>
-2005-10-11 Julien MOUTTE <julien@moutte.net>
+ * tests/check/generic/states.c:
+ plug test leak
+ Original commit message from CVS:
+ plug test leak
- * 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.
+2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-10-11 Wim Taymans <wim@fluendo.com>
+ 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
- * check/generic/states.c: (GST_START_TEST):
- remove old property.
+2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2005-10-11 Michael Smith <msmith@fluendo.com>
+2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * 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.
-2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * 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.
+2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ 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).
- * ext/alsa/gstalsasink.c:
- Also allow unsigned int.
+2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Small cleanup
+ 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.
-2005-10-10 Wim Taymans <wim@fluendo.com>
+2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Small update, use API as stated in design docs.
+ 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).
- * 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.
+2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * 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
- * 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.
+2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-09 Philippe Khalaf <burger@speedy.org>
+ 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-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
+2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+ 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/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
+2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
- (plugin_init):
- Add wavpack and spc typefind functions from 0.8 branch.
+2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+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).
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ gst-libs/gst/audio/audio.c: Fix documentation.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Fix documentation.
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
- * 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.
+ 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.
-2005-10-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
- Fix leaks in typefind registration
- Clean up the gratuitous commenting and whitespacing a little
+ 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)
-2005-10-08 Wim Taymans <wim@fluendo.com>
+2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
- Only actually wait for the thread to be stopped if it's
- running.
+ 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.
-2005-10-08 Wim Taymans <wim@fluendo.com>
+2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.spec.in:
+ add header file for easy codec install
+ Original commit message from CVS:
+ add header file for easy codec install
- * 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.
+2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- Report the FLOW_RETURN as string in the error message.
+2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
- Don't assert when clearing an unnegotiated buffer.
+ 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.
-2005-10-04 Michael Smith <msmith@fluendo.com>
+2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
- * 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.
+ 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).
-2005-10-04 Michael Smith <msmith@fluendo.com>
+2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.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/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.
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
- * configure.ac:
- back to development
+ 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.
-=== release 0.9.3 ===
+2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- Releasing 0.9.3, "De Facto"
+2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-03 Andy Wingo <wingo@pobox.com>
+ * 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).
- * gst/playback/gstdecodebin.c (try_to_link_1)
- (remove_element_chain): set element to NULL before removing it.
+2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-10-02 Johan Dahlin <johan@gnome.org>
+ 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).
- * ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_uri_get_protocols):
- protect gst_gnomevfs_get_supported_uris by a mutex, to make it
- MT safe.
+2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
-2005-10-02 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-29 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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.
-2005-09-29 Wim Taymans <wim@fluendo.com>
+2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- We use fixed caps.
+ 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/playback/Makefile.am:
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- Added example stream introspection code.
+2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
-2005-09-28 Stefan Kost <ensonic@users.sf.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/adder/gstadder.c: (gst_adder_collected):
- fix adder for float elements
+2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-28 Wim Taymans <wim@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.
- * 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
+2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-28 Andy Wingo <wingo@pobox.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.
- * gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
- and unlocking.
+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.
- * gst/sine/gstsinesrc.c (gst_sinesrc_unlock): Actually implement
- unlocking.
+2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
- Actually add the pad template.
- (gst_tcpclientsink_get_type): We're a base sink. Woot, works.
+ 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.
- * gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
- I'm at it...
+2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
- * gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
- Proper EOS handling.
+2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-27 Andy Wingo <wingo@pobox.com>
+ 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.
- * gst/tcp/gsttcpclientsrc.c: Cleaned up.
+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.
- * gst/tcp/gsttcpserversrc.c: Cleaned up.
+2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
+ 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).
- * 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.
+2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
+ 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.
-2005-09-26 Andy Wingo <wingo@pobox.com>
+2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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/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/volume/gstvolume.c (volume_transform_ip): Update for
- controller api change.
+2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
-2005-09-24 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * 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
+2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-09-23 David Schleef <ds@schleef.org>
+ 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
- * gst/audioresample/Makefile.am:
- * gst/audioresample/debug.h:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/resample.c: Convert to using gst debugging
+2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_send_event):
- Only seek on one sink, the first one that succeeds.
+2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-09-22 Michael Smith <msmith@fluendo.com>
+ 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.
- * 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.
+2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-21 Andy Wingo <wingo@pobox.com>
+ 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.
- * gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
- thingies.
+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/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
- can be called multiple times, dogs.
+2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-09-21 Wim Taymans <wim@fluendo.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-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.
+2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-21 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * docs/libs/compiling.sgml:
- fixing typos
+2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-09-20 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-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.
+2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-20 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-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.
+2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- fix wrong pop/unref
+2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-09-19 Torsten Schoenfeld <kaffeetisch at gmx dot 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.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.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)
+ 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.
-2005-09-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/rtp/gstrtpbuffer.h: Add some more fixed payloads.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Add some more fixed payloads.
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * examples/seeking/seek.c: (make_wav_pipeline), (main):
- Fixed wav pipeline.
-
-2005-09-19 Wim Taymans <wim@fluendo.com>
-
- * 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.
+ 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).
-2005-09-19 Wim Taymans <wim@fluendo.com>
+2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-09-19 Wim Taymans <wim@fluendo.com>
+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.
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Fix for bus API.
+2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.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_set_gst_timestamp),
- (gst_base_rtp_depayload_queue_release):
- Some cleanups.
+ 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.
- * 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.
+2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-18 David Schleef <ds@schleef.org>
+ 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).
- * 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)
+2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * check/elements/audioconvert.c: (GST_START_TEST):
- try out 24 bit conversion
+2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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.
+2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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 :)
+2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Wim Taymans <wim@fluendo.com>
+ 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).
- * gst/audioconvert/gstaudioconvert.c:
- And enable 24 bits mode as well..
+2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-15 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-09-15 Andy Wingo <wingo@pobox.com>
+ 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.
- * configure.ac (plugindir): Remove the EOL matcher from the
- regexp, as it causes me problems. Libtool? Make? Who knows?
+2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-09-14 David Schleef <ds@schleef.org>
+ 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).
- * 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.
+2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ 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.
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/theora/theoraenc.c:
- add theoraenc
+ * 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.
-2005-09-13 Tim-Philipp Müller <tim at centricular dot net>
+2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
- * 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)
+ * gst-plugins-base.spec.in:
+ add latest files
+ Original commit message from CVS:
+ add latest files
- * gst/playback/test4.c: (main):
- Fix MinGW build problem and use g_usleep() instead of
- sleep() (#316162)
+2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-12 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/typefind/gsttypefindfunctions.c: Fix comment.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ Fix comment.
- * 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
+2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-09 Stefan Kost <ensonic@users.sf.net>
+ 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).
- * 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
+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.
-2005-09-08 Jan Schmidt <thaytan@mad.scientist.com>
+2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
+2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+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.
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- gsttaginterface.h -> gsttagsetter.h
+2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+ 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).
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- added another test that failes for me (test is not active by default)
+2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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).
- * configure.ac:
- v4l2 is no longer in gst-plugins-base
+2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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>
+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.
- * ext/libvisual/Makefile.am:
- link against gst-base-libs
+2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
-2005-09-06 David Schleef <ds@schleef.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>
- * configure.ac: Enable libvisual plugin.
- * ext/libvisual/Makefile.am:
- * ext/libvisual/visual.c: Fixes to make it compile.
+ 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.
-=== release 0.9.2 ===
+2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-09-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * NEWS:
- * RELEASE:
- * configure.ac:
- * docs/random/ChangeLog-0.8:
- releasing 0.9.2, "Spoon"
+2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-05 Michael Smith <msmith@fluendo.com>
+ 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).
- * 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.
+2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
- fix distcheck
- * gst/audioresample/resample.c:
- fix wrong docstring
+2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * common/gst-xmlinspect.py:
- * common/gtk-doc-plugins.mak:
- only inspect plugins for this given package
- require gst-python 0.9
+2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * 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
+2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
-2005-08-31 Wim Taymans <wim@fluendo.com>
+ 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-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Resync if the buffer timestamps drift more than a 10th
- of a second.
+2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-31 Tim-Philipp Müller <tim at centricular dot net>
+ Dist design docs.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/design/Makefile.am:
+ Dist design docs.
- * 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).
+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
-2005-08-30 Wim Taymans <wim@fluendo.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)
- * check/elements/audioconvert.c: (setup_audioconvert):
- * check/elements/audioresample.c: (setup_audioresample):
- * check/elements/volume.c: (setup_volume):
- Fix checks.
+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
-2005-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * common/gtk-doc-plugins.mak:
- * common/plugins.xsl:
- * docs/plugins/Makefile.am:
- make module a param
+ 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.
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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).
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform):
- do not update controlled params, if buffer has no timestamp
+ 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.
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- controllerized elements also need to link against controller-libs ;)
+ 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.
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2005-08-29 Andy Wingo <wingo@pobox.com>
+2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push)
- (vorbis_handle_data_packet): Fix some int overflow errors.
+ add doap file
+ Original commit message from CVS:
+ * Makefile.am:
+ * gst-plugins-base.doap:
+ * gst-plugins-base.spec.in:
+ add doap file
- * 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.
+2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
- * check/elements/volume.c (cleanup_volume): Fix for running
- CK_FORK=no.
+ 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.
- * check/elements/audioconvert.c: Convert from native endian, not
- little endian.
+2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-08-29 Michael Smith <msmith@fluendo.com>
+ configure.ac: Bump version nano - back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version nano - back to CVS.
- * 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>
+=== release 0.10.11 ===
- * gst/audioresample/gstaudioresample.c:
- use base class' newsegment to properly timestamp
+2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-08-26 Wim Taymans <wim@fluendo.com>
+ 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"
- * 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.
+2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-08-26 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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.
+2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * po/.gitignore:
+ Ignore all pot files
+ Original commit message from CVS:
+ Ignore all pot files
- * 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!
+2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose),
- (audio_convert_get_unit_size):
- plug some leaks
+2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * 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
+2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-08-25 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
- * 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
+2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
-2005-08-25 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- Revert unpopular change for GST_MESSAGE_SRC to GObject.
+ 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.
-2005-08-25 Stefan Kost <ensonic@users.sf.net>
+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/volume/gstvolume.c:
- made set_caps function static
+2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_change_state):
- Stop leaking taglists.
+2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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/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.
+2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
- Basesink does newsegment calculations for us now.
+2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-24 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.
- * 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
+2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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.
+2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ 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
- * configure.ac:
- Remove audioscale.
+2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * 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
+2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+ 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).
- * check/Makefile.am:
- Add CHECK_CFLAGS and LDFLAGS
+2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- GST_MESSAGE_SRC became a GObject
+ 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.
-2005-08-24 Wim Taymans <wim@fluendo.com>
+2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-08-24 Andy Wingo <wingo@pobox.com>
+2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/v4l/gstv4lelement.c (gst_v4lelement_start)
- (gst_v4lelement_stop): Call _start and _stop for xoverlay instead
- of _open and _close.
+ 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/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:
+2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
-2005-08-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * examples/seeking/seek.c: (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
- Small seek updates.
+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.
-2005-08-23 Andy Wingo <wingo@pobox.com>
+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-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Only fixate endianness if it is
- present in the caps.
+2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
-2005-08-22 Andy Wingo <wingo@pobox.com>
+ 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).
- * ext/alsa/gstalsasink.c (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
- device-name property.
+2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
- close_device in the ring buffer, like gstaudiosink.
+ 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.
- * 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.
+2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
- * 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
+ 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.
-2005-08-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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>
+ 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
- * 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.
+2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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>
+ 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.
- * 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.
+2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
-2005-08-18 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
- moved statement below switch
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- added debug ptr
+2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
-2005-08-16 Wim Taymans <wim@fluendo.com>
+ 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-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Open and close device in READY<->NULL state change.
+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).
-2005-08-16 Andy Wingo <wingo@pobox.com>
+2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
- * examples/seeking/Makefile.am: Don't compile non-compiling
- compiled objects with the compiler.
+ 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.
- * examples/seeking/seek.c (make_dv_pipeline): Update for new DV
- elements.
+2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
-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.
+ 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.
-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.
+2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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!)
+2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-08 Edward Hervey <edward@fluendo.com>
+ 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).
- * gst-libs/gst/riff/riff-media.c:
- backported updates since branch
+2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-08 Andy Wingo <wingo@pobox.com>
+ gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
+ Another typo fix (#366212).
- * 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.
+2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.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/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.
- * 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.
+2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
- (gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
- base class API.
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
- * gst-libs/gst/audio/gstringbuffer.h
- (GstRingBufferClass.open_device, GstRingBufferClass.close_device):
- New vmethods.
+2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
- * 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.
+ 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.
-2005-08-08 Tim-Philipp Müller <tim at centricular dot net>
+2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/interfaces/mixer.h:
- Reset padding to GST_PADDING.
+ 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).
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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-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.
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * ext/alsa/gstalsasink.c: (set_hwparams):
- Assign debug category, add negotiation debug msgs.
+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.
-2005-08-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_start):
- Fix error code for file-not-found to NOT_FOUND.
+ 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.
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.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
+ 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).
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/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.
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c: (gen_video_element):
- Enable videoscale.
+ 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)
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/gconf/gconf.c:
- * gst-libs/gst/gconf/gconf.h:
- Fix some Andy Problem [tm].
+ 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.
-2005-08-04 Andy Wingo <wingo@pobox.com>
+2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
- * gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c
- (gst_ffmpegcsp_get_size): Adapt to API changes.
+ 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.
- * gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
- Implement an in-place do-nothing transform.
+2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-03 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-02 Edward Hervey <edward@fluendo.com>
+ 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/volume/gstvolume.c:
- include left from controller cleanup
+2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-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.
+ 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.
-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.
+2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-08-01 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * 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)
+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.
-2005-07-29 Wim Taymans <wim@fluendo.com>
+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/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
- Convert me to BaseTransform!! help..
+2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-29 Andy Wingo <wingo@pobox.com>
+ 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.
- * ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
- sinks.
+2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
- support of both endiannesses.
+ 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).
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix confusing debug message (s/event/query/)
+ 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.
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+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.
- * gst/videotestsrc/videotestsrc.h:
- Use "_stdint.h" instead of <stdint.h>
+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).
-2005-07-27 Wim Taymans <wim@fluendo.com>
+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).
- * ext/vorbis/Makefile.am:
- Revert wrong commit.
+2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
-2005-07-27 Wim Taymans <wim@fluendo.com>
+ * 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/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
- More compilation fixen.
+2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-27 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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
+2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
-2005-07-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/videoscale/gstvideoscale.c: (gst_videoscale_prepare_images):
- Fix segfault for I420/YV12.
+2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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)
- * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
- Report bitrate.
+2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-07-21 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/playback/gstplaybasebin.c: (gen_preroll_element):
- * gst/playback/gstplaybin.c: (setup_sinks),
- (gst_play_bin_change_state):
- Refcount fix and more comments.
+2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
-2005-07-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * 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.
+2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- Use _new_custom() so we can set custom message types for buffering
- messages.
+ 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.
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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
-2005-07-20 Wim Taymans <wim@fluendo.com>
+2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-07-19 Andy Wingo <wingo@pobox.com>
+2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/videotestsrc/: Cleanups.
+ 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
-2005-07-19 Wim Taymans <wim@fluendo.com>
+2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Better debugging.
+ * 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.
-2005-07-19 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-18 Andy Wingo <wingo@pobox.com>
+ 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.
- * 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.
+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.
-2005-07-18 Wim Taymans <wim@fluendo.com>
+2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-07-18 Wim Taymans <wim@fluendo.com>
+2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ tests/check/pipelines/oggmux.c: Remove obsolete comment.
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c:
+ Remove obsolete comment.
- * 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.
+2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
-2005-07-16 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Align samples even if we have roundoff errors in the
- timestamp conversion.
+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?)
-2005-07-16 Wim Taymans <wim@fluendo.com>
+2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
- * 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.
+2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2005-07-16 Wim Taymans <wim@fluendo.com>
+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>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_buffer):
- Timestamp buffers correctly.
+ * gst-plugins-base.spec.in:
+ add new header file to spec
+ Original commit message from CVS:
+ add new header file to spec
- * gst/playback/gstplaybin.c: (gen_video_element):
- Make internal fakesink silent.
+2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-15 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_transform):
- * gst/volume/gstvolume.c: (volume_transform):
- Ported to new API.
+ 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
-2005-07-14 Wim Taymans <wim@fluendo.com>
+2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
- Also draw image when not from a pool.
-
-2005-07-14 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
- (mute_stream), (silence_stream):
- Small debug additions.
-
-2005-07-14 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2005-07-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2005-07-13 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2005-07-13 Andy Wingo <wingo@pobox.com>
+ 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.
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_stop): Fix a spurious warning.
- (gst_v4lsrc_fixate): Fixate on format as well.
+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.
- * 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.
+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>
-2005-07-12 Andy Wingo <wingo@pobox.com>
+ * ChangeLog:
+ ChangeLog surgery for missing bug-number
+ Original commit message from CVS:
+ ChangeLog surgery for missing bug-number
- * 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.
+2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
-2005-07-11 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).
- * 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
+2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-10 Thomas Vander Stichele <thomas at apestaart dot 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.
- * 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>
+2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/: 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.
- * 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.
+2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/v4l_calls.h: Cast to V4lElement.
- * sys/v4l/v4l_calls.c: Header loc fixen, don't load mjpeg, all
- v4lelements are sources.
+ 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.
- * 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.
+2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lelement.c: Derive from GstPushSrc. No more
- signals. Some cleanups.
+ 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).
- * sys/v4l/gstv4lcolorbalance.h: Interface header update.
+2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4l.c: Don't register v4lelement, or the jpeg/mjpeg
- stuff.
+ 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/v4l/Makefile.am: Build everything except the jpeg/mjpeg
- stuff.
+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.
- * sys/Makefile.am (SUBDIRS): Hit the V4L crack pipe.
+2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-07-07 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
-2005-07-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-06 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.
- * ext/theora/theoradec.c: (theora_dec_src_getcaps),
- (theora_dec_push), (theora_handle_data_packet):
- Prepare for better timestamp fix later.
+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.
- * gst/audioconvert/gstaudioconvert.c:
- List most accurate caps first
+2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
- Use proper pad task function.
+ * 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)
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_show_frame):
- Fix deadlock when alloc failed.
+2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-07-05 Andy Wingo <wingo@pobox.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/gnomevfs/gstgnomevfssrc.c:
- * gst/sine/gstsinesrc.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/tcp/gsttcpclientsrc.c: s/BASESRC/BASE_SRC/.
+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
- * sys/v4l/: Port from 0.8.
+2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Many files: Null if we got it....
+ * 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
-2005-07-05 Andy Wingo <wingo@pobox.com>
+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/tcp/gsttcpserversink.c (gst_tcpserversink_handle_server_read):
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_client_queue_data):
- Signedness fixes.
+2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-07-05 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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>
+2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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()
-2005-07-01 Andy Wingo <wingo@pobox.com>
+2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
- * gst/videoscale: Merge David's work from 0.8 branch. Changes to
- come later.
+ 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.
-2005-06-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ 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
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Fix build after riff changes.
+ 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
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init):
- Oops, I shouldn't apply hacks.
+ 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.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init):
- Remove pad_loop function which doesn't work.
+ 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.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/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.
-2005-06-29 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+=== release 0.10.10 ===
-2005-06-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * autogen.sh:
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
* 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
-
-2005-06-28 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_init):
- Removed pad loop function.
-
-2005-06-28 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2005-06-28 Andy Wingo <wingo@pobox.com>
-
- * *.c: Don't cast to GstObject before reffing/unreffing.
-
-2005-06-27 Andy Wingo <wingo@pobox.com>
-
- * gst/videotestsrc/gstvideotestsrc.c
- (gst_videotestsrc_activate_push): Activation API changes.
-
- * 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.
-
- * 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.
-
-2005-06-27 Edward Hervey <edward@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_dec_change_state):
- re-arranged call to parent's state change in order to avoid locks (or
- worse).
-
-2005-06-26 Edward Hervey <edward@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- 2nd argument of 'unknow-type' signal is a GstCaps and not a
- GstMiniObject
-
-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.
-
-2005-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
+ * 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
-2005-06-23 Christian Schaller <uraeus@gnome.org>
+2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: add videorate
- * gst-plugins-base.spec.in: add videorate
+ * configure.ac:
+ * win32/common/config.h:
+ second prerelease
+ Original commit message from CVS:
+ second prerelease
-2005-06-23 Wim Taymans <wim@fluendo.com>
+2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * ChangeLog:
+ update bug in changelog
+ Original commit message from CVS:
+ update bug in changelog
-2005-06-23 Wim Taymans <wim@fluendo.com>
+2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
- * ext/ogg/README:
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_set_header_on_caps):
- Buffer on caps is not boxed anymore.
+ 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'
-2005-06-22 Wim Taymans <wim@fluendo.com>
+2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
- * 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.
+ 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
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
- back to HEAD
-
-=== release 0.9.1 ===
+ * win32/common/config.h:
+ first prerelease
+ Original commit message from CVS:
+ first prerelease
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * .cvsignore:
- * NEWS:
- * README:
- * RELEASE:
- * configure.ac:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
- updates for release
+ update po files
+ Original commit message from CVS:
+ update po files
-2005-06-09 Andy Wingo <wingo@pobox.com>
+2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
-
-2005-06-09 Andy Wingo <wingo@pobox.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.
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/net/Makefile.am:
- Add gstnet to build.
+2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-09 Andy Wingo <wingo@pobox.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-libs/gst/gconf/gconf.c:
- * gst/playback/test.c:
- * gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
- fixes.
+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.
- * gst/audioconvert/gstaudioconvert.c: RPAD fixes.
+2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
- * ext/theora/theoraenc.c (theora_enc_chain):
- * ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.
+ 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
- * ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
- RealPad.
+2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-02 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst-libs/gst/net/Makefile.am:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs.pc.in:
- Added net stuff, version net lib.
-
-2005-06-02 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_vorbis_theora_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (do_seek):
- Updated seek example.
+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.
-2005-06-02 Andy Wingo <wingo@pobox.com>
+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").
- * pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
- * pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
- list.
+2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.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/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).
-2005-06-02 Wim Taymans <wim@fluendo.com>
+2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.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/adder/gstadder.c: Fix build.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_src_event):
+ Fix build.
- * 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>
+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).
- * 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.
+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.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Fix memleaks, GST_BUFFER_DATA() is not freed.
+ 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.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Open non-blocking, set to blocking mode afterwards to avoid
- lockups when audio device is busy.
+ 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.
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
- This can't be good.
+ 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
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-28 16:17:13 +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.
+ 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.
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-28 16:09:57 +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).
+ * 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.
-2005-05-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * examples/Makefile.am:
- fix buildbot (make distcheck)
+ 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.
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybin.c: (gen_vis_element):
- Remove some wrong code. Doesn't work yet.
+ 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.
-2005-05-19 Wim Taymans <wim@fluendo.com>
+2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
- Don't lock an unassigned variable.
+ 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.
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-08-23 19:37:50 +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.
+ tests/check/elements/.cvsignore: make buildbot happy
+ Original commit message from CVS:
+ * tests/check/elements/.cvsignore:
+ make buildbot happy
-2005-05-18 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-05-17 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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:
+2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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>
+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.
- * 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>
+2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Disable libvisual
+ 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).
- * 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>
+2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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:
+ 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.
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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..
+ 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.
-2005-05-06 Christian Schaller <uraeus@gnome.org>
+2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * pkgconfig/gstreamer-libs.pc.in: add missing library calls
+ * gst/gdp/README:
+ adding a README
+ Original commit message from CVS:
+ adding a README
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.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.
+ 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.
+2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-05-06 Wim Taymans <wim@fluendo.com>
+ 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().
- * 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.
+2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-05-06 Zaheer Abbas Merali <zaheerabbas at merali 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).
- * ext/Makefile.am:
- Disable cdparanoia until someone ports it!
+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
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_sink_activate):
- And revert after wingo's revert.. sigh..
+ 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
-2005-05-05 Andy Wingo <wingo@pobox.com>
+2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ * 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.
-2005-05-05 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-05-05 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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>
+2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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>
-
- * 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.
+ 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().
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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).
-2005-04-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gst/audioconvert/Makefile.am:
- * gst/audioscale/Makefile.am:
- Fix part of the build. Come on guys, autogen didn't even work :)
+ 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.
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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).
+2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.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).
- * 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).
+2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-04-25 David Schleef <ds@schleef.org>
+ 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"
- * 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:
+2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-04-25 David Schleef <ds@schleef.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.
- 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:
+2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
- Convert to 0.9 API, seems to work:
- * sys/ximage/Makefile.am:
- * sys/ximage/ximagesink.c:
+ 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).
-2005-04-24 David Schleef <ds@schleef.org>
+2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
- Link plugins against libraries:
- * ext/alsa/Makefile.am:
- * gst/tcp/Makefile.am:
+ 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).
- Remove asm code that should be in liboil
- * gst/videoscale/Makefile.am:
- * gst/videoscale/videoscale_x86_asm.s:
+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.
- 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:
+2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-24 David Schleef <ds@schleef.org>
+ 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).
- 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):
+2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- Link plugins against libraries:
- * ext/libvisual/Makefile.am:
- * sys/xvimage/Makefile.am:
+ 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).
-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:
+2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
- Create proper libraries:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
+ 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).
- 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>
-
- * 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.
+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).
-2005-04-20 Wim Taymans <wim@fluendo.com>
+2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/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.
- * 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.
+2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.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.
-2005-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
- * autogen.sh:
- * configure.ac:
- * ext/Makefile.am:
- * gst/Makefile.am:
- * po/POTFILES.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:
- * 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.
+ 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.
-2005-02-22 Thomas Vander Stichele <thomas at apestaart dot org>
+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
- * configure.ac:
- hunting season on 0.9 is now OPEN
+2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-02-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Kick the hell out of gcc for not warning me about a symbol conflict.
+2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-02-22 Luca Ognibene <luogni@tin.it>
+ * 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.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+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.
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_link):
- Don't leak caps string (fixes #168134)
+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.
- * 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).
+2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-02-21 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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).
+2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-21 Maciej Katafiasz <mathrick@freedesktop.org>
+ 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
- * 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.
+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).
-2005-02-21 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain):
- Make sure we only write to writable buffers
+2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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.
+2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.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)
+ 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
-2005-02-19 Martin Holters <martin.holters@gmx.de>
+2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ 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
- * gst/audioconvert/bufferframesconvert.c:
- (buffer_frames_convert_link):
- Don't leak othercaps. (fixes #167878)
+2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-19 Arwed v. Merkatz <v.merkatz@gmx.net>
+ 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.
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_srclink),
- (gst_visual_change_state):
- Support libvisual 0.2.0.
+2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-18 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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.
+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
-2005-02-17 Tim-Philipp Müller <tim at centricular dot net>
+=== release 0.10.9 ===
- * 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.
+2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-02-17 Luca Ognibene <luogni at tin dot it>
+ 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"
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send):
- Don't pass uninitialised values to setsockopt(). (fixes #167704)
+ 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.
-2005-02-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybin.c: (add_sink):
- Invert bin_add/link order to workaround deadlock in opt.
+ * 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>
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/modplug/gstmodplug.cc:
- Add missing break causing position queries to fail.
+2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate):
- Granpos can apparently be -1, which screws up calculations...
+2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-16 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * 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
+2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-13 Benjamin Otte <otte@gnome.org>
+ 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.
- * 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)
+2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
-2005-02-12 Tim-Philipp Müller <tim at centricular dot net>
+ * 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).
- * 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>
+2006-07-06 13:12:02 +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/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.
-2005-02-09 David Schleef <ds@schleef.org>
+2006-07-06 13:04:24 +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.
+ 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.
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
- * ext/mplex/gstmplexibitstream.cc:
- gcc madness.
+ 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.
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstogmparse.c:
- * gst/debug/gstnavigationtest.c:
- Die, thou faulty symbol pollutors (non-static parent_class).
+ ext/alsa/gstalsamixertrack.c: Some const-ification.
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
+ Some const-ification.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/mplex/gstmplexibitstream.cc:
- Fix event handling (#165525).
+ 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.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/mikmod/gstmikmod.c:
- * gst/modplug/gstmodplug.cc:
- Add missing endianness to template (fixes #165509).
+ ext/vorbis/vorbisdec.c: Remove unused variable.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
+ Remove unused variable.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_data):
- Fix wrong order of reading of optional bytes (#165290).
+ Makefile.am: include lcov.mak
+ Original commit message from CVS:
+ * Makefile.am:
+ include lcov.mak
+ * configure.ac:
+ add GCOV_LIBS to GST_LIBS
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Implement FILLER event awareness.
+ 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.
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_convert):
- Fix track calculations (#166208).
+ 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).
-2005-02-08 Gergely Nagy <algernon@bonehunter.rulez.org>
+2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * ext/libpng/gstpngdec.c: (gst_pngdec_init), (gst_pngdec_chain):
- * ext/libpng/gstpngenc.c:
- Fix byte-order, use proper fixed caps. Fixes #164197.
+2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+ 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
- * configure.ac:
- Add dvdlpcmdec
+2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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>
+ 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/subparse/gstsubparse.c:
- Fix OSX buildbot.
+2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * 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.
+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.
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/mpegstream/gstdvddemux.c:
- Add audio/x-dts to audio pad template caps
+ 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.
-2005-01-30 David Schleef <ds@schleef.org>
+2006-06-22 11:01:41 +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.
+ 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'.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
- Add 3IV2 fourcc.
+ 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).
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-22 10:31:22 +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/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.
-2005-01-29 Ronald S. 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).
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- D'oh, reference the palette data, not the palette structure.
- Fixes color distortion in #132341.
+2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- PAR can be non-fixed when not provided as argument (#162626).
+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>
-2005-01-29 David Moore <dcm@acm.org>
+ * gst-plugins-base.spec.in:
+ update for latest changes
+ Original commit message from CVS:
+ update for latest changes
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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>.
+ 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
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
- * gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_init):
- Use the src template for creating the src pad (#162330).
+ 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
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-17 14:18:41 +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>
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
- * ext/Makefile.am:
- Unbreak buildbot.
+2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-28 Andy Wingo <wingo@pobox.com>
+ 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/dv/gstdvdec.c: Change the pixel aspect ratio of dvdec output
- to reflect a different dubious internet source. Add a reference
- and some commentary.
+2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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>
+2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
- * 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>
+ 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/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.
+2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-25 Edward Hervey <bilboed@bilboed.com>
+ 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
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/effectv/gstquark.c: (gst_quarktv_class_init),
- (gst_quarktv_change_state), (gst_quarktv_dispose):
- Memory free'ing location fix (#164708).
+ * 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)
-2005-01-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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>
-2005-01-25 Andy Wingo <wingo@pobox.com>
+ 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).
- * sys/v4l/gstv4lelement.c (gst_v4l_iface_supported): Fix compile
- for #ifndef HAVE_XVIDEO.
+2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-24 Jeffrey C. Ollie
+ 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.
- reviewed by: Maciej Katafiasz <mathrick@freedesktop.org>
+2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-01-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Audio can be <8000Hz.
+ 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).
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Explicit state change to workaround refcount bugs.
+ 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.
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavimux.c: (gst_avimux_write_tag),
- (gst_avimux_riff_get_avi_header):
- Fix...
+ 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).
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
- * 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.
+ 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).
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Safety guard.
+ po/POTFILES.in: add missing files containing translatable strings
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ add missing files containing translatable strings
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/avi/gstavimux.c: (gst_avimux_write_tag):
- Fix padding...
+ 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
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/matroska/ebml-read.c: (gst_ebml_read_buffer):
- Allow for 0-sized buffers. Fixes length query problems in
- starwars.mkv from the testsuite.
+ ignore more
+ Original commit message from CVS:
+ * docs/libs/.cvsignore:
+ * tests/check/elements/.cvsignore:
+ * tests/check/libs/.cvsignore:
+ ignore more
-2005-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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>
+ 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/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.
+2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add intel-h263.
+2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * 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.
+2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
- Elements may already be destroyed when this function is called.
+2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * common:
+ * win32/common/config.h:
+ update to cvs
+ Original commit message from CVS:
+ update to cvs
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (gst_qtdemux_handle_esds):
- More memory leak fixes (#149162).
+2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_add_stream):
- Fix two memleaks.
+2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
- Argh...
+2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * configure.ac:
+ back to trunk
+ Original commit message from CVS:
+ back to trunk
- * 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.
+=== release 0.10.8 ===
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/dvdread/dvdreadsrc.c: (get_next_cell_for):
- Fix compile warnings on Solaris 10 buildbot
+ * 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
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/dvdread/dvdreadsrc.c: (_read):
- Don't read beyond the last cell in a chapter (fixes
- invalid memory access)
+ 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
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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)
+ 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
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix BGRA32 caps (#164209).
+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.
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. 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
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- alpha_mask can be RGBA/ABGR. Fixes #164265.
+2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
-2005-01-17 Francis Labonte <francis_labonte@hotmail.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
+ 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.
-2005-01-17 Steve Baker <steve@stevebaker.org>
+2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (aiff_type_find),
- (svx_type_find), (sds_type_find), (ircam_type_find), (plugin_init):
- Add libsndfile typefind functions (#163309).
+2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/audioconvert.c: More correct float->int conversion.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float):
+ More correct float->int conversion.
- * tools/gst-launch-ext-m.m:
- Add .aac, fix .m1v/.m2v (#163891).
+2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+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>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/playback/gstplaybasebin.c: (probe_triggered):
- Of course, only pause if group is done...
+2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Thread safety.
+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
-2005-01-16 Jan Schmidt <thaytan@mad.scientist.com>
+2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
- * ext/swfdec/gstswfdec.c: (gst_swfdec_change_state):
- Don't return state change success when the parent
- failed.
+ 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
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavimux.c: (gst_avimux_handle_event):
- Free events (fix memleak in #162905).
+ 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.
-2005-01-15 Gergely Nagy <algernon@bonehunter.rulez.org>
+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
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- Fix for depth = 15. Fixes #161675.
+ * 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.
-2005-01-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-01-14 Ronald S. 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).
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Revert Johan's 1.35->1.36 since it breaks compat.
+2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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
+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.
-2005-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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.
-2005-01-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
- Fix for if items are already in list...
+ 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.
-2005-01-12 Benjamin Otte <otte@gnome.org>
+2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/adder/gstadder.c: (gst_adder_loop):
- fix adder a bit so it doesn't screw up with events as much anymore
+ * 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).
-2005-01-12 Jan Schmidt <thaytan@mad.scientist.com>
+2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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: 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.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/avi/gstavimux.c: (gst_avimux_class_init),
- (gst_avimux_pad_unlink), (gst_avimux_release_pad):
- Reusability fixes.
+ 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
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.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 flags when requested.
+ 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.
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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_rates_probe):
- Fix dmix.
+2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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
+2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Ronald S. 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/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.
+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.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-18 19:21:53 +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
+ 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).
-2005-01-10 Ronald S. 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>
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Don't crash, biatch! :).
+2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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...
+2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
- Also shove tags on kid pads.
+2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+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.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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.
+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.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event):
- Fix warning (#161191).
+ 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).
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dvdread/stream_labels.c:
- (dvdreadsrc_get_audio_stream_labels):
- Fix warning (init the good variable in switch default)
+ 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).
-2005-01-09 Koop Mast <kwm@rainbow-runner.nl>
+2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/tta/gstttaparse.c: (gst_tta_src_event):
- Fix gcc-2.95 compile (#163485).
+2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+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).
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybasebin.c: (group_destroy):
- Remove hack to get rid of assert and get rid of unlinked
- signals properly.
+ * win32/common/config.h:
+ update config.h
+ Original commit message from CVS:
+ update config.h
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * ext/ogg/README:
+ fix mistakes in README
+ Original commit message from CVS:
+ fix mistakes in README
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
- * testsuite/embed/Makefile.am:
- test->noinst, fix make test in buildbot.
+ 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.
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.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.
+ * configure.ac:
+ Back to CVS
+ Original commit message from CVS:
+ Back to CVS
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+=== release 0.10.7 ===
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix hanging subs.
+2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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"
- * 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>.
+2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * testsuite/embed/Makefile.am:
- Fix buildbot.
+2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-09 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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)
- * 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).
+2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ win32/MANIFEST: update win32 files listing
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ update win32 files listing
- * gst-plugins.spec.in:
- Add subparse.
+2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * configure.ac:
- Since we use functions from CVS, up requirement.
+2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * 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>
+2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
-2005-01-07 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
- * 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
+ 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
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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).
- * 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.
+2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-07 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * gst/tcp/gstmultifdsink.c:
+ more commenting
+ Original commit message from CVS:
+ more commenting
-2005-01-07 Sebastien Cote <sc5@hermes.usherb.ca>
+2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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).
+2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Fix against template (#150576).
+2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-06 Benjamin Otte <otte@gnome.org>
+ 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.
- * gst/games/gstpuzzle.c: (draw_puzzle):
- don't draw a puzzle if either width or height of tiles would be 0.
+2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-06 Benjamin Otte <otte@gnome.org>
+ * tests/check/elements/multifdsink.c:
+ set caps and plug leaks
+ Original commit message from CVS:
+ set caps and plug leaks
- * 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
+2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-06 Benjamin Otte <otte@gnome.org>
+ 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
- * gst/games/gstvideoimage.c: (copy_hline_YUY2):
- fix stupid typo that borked copying on YUY2
+2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-06 Benjamin Otte <otte@gnome.org>
+ 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
- * gst/games/gstpuzzle.c: (draw_puzzle):
- fix edges when image sizes aren't multiples of tile sizes
+2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-06 Benjamin Otte <otte@gnome.org>
+ * gst/tcp/gstmultifdsink.c:
+ DEBUG_FUNCPTR'ing
+ Original commit message from CVS:
+ DEBUG_FUNCPTR'ing
- * 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
+2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ * 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>
- * 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)
+ 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.
-2005-01-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_chain):
- Only increment timestamp if it's valid. Fixes raw AAC streams.
+ 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.
-2005-01-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
- * 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.
+ 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).
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ 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
-2005-01-06 Paul Jack <pjack@sfaf.org>
+2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
- Fix segfault (#161667).
+2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-01-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Fix framerate reporting.
+2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-05 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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>
+2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- back to cvs
+ 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.
-=== release 0.8.7 ===
+2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.7, "Hyperspace"
+2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstplaybasebin.c:
- Fix for #162924 - free caps after use, not before
+ 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
-2005-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst/playback/gstplaybasebin.c:
- * gst/wavparse/gstwavparse.c:
- Fix for #154773 - fixes playback of small .wav files
+2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioscale/gstaudioscale.c:
- Fix for #162819 - make audioscale reusable
- Fixes playback of more than one file with playbin/totem
+ 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
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.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
+ 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.
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
- * gst/ffmpegcolorspace/imgconvert.c:
- give some indication of why we're segfaulting
+ * 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.
-2004-12-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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>
+ 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>
- * 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/rtp/gstbasertppayload.c:
+ some RTP debug
+ Original commit message from CVS:
+ some RTP debug
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioconvert/gstchannelmix.c:
- fix for GLIB < 2.4
+ 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.
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Makefile.am:
- * configure.ac:
- disable docs again until it actually passes make distcheck.
+ 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.
-2004-12-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * Makefile.am:
- really fix dist
+ * tests/check/elements/audioconvert.c:
+ dump expected data when audioconvert test fails
+ Original commit message from CVS:
+ dump expected data when audioconvert test fails
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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...
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theoraenc.c:
+ add another include
+ Original commit message from CVS:
+ add another include
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- Put additional LAME check inside the conditional. Fixes #152339
+ * gst/subparse/gstssaparse.c:
+ atoi() needs stdlib.h
+ Original commit message from CVS:
+ atoi() needs stdlib.h
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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/playback/test4.c:
+ * gst/playback/test5.c:
+ * gst/playback/test6.c:
+ exit needs stdlib.h
+ Original commit message from CVS:
+ exit needs stdlib.h
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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>
-2004-12-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- Fix up.
-
-2004-12-22 Archana Shah <archana.shah@wipro.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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
-
-2004-12-22 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
-
- * Makefile.am: Make sure docs gets disted
- * docs/Makefile.am: Make sure all needed files get disted
- * gst-plugins.spec.in: latest updates
-
-2004-12-22 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-12-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * sys/oss/README: remove this file, which predates my birth
- (and which content is by far outdated)
-
-2004-12-20 Stefan Kost <ensonic@users.sf.net>
-
- * Makefile.am:
- * configure.ac:
+ * ChangeLog:
+ * common:
* 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/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/Makefile.am:
* docs/upload.mak:
- * docs/version.entities.in:
- Added boilerplate gtk-doc files for plugin-libs documentation.
+ use common upload.mak
+ Original commit message from CVS:
+ use common upload.mak
-2004-12-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/auparse/gstauparse.c: fix int and float audio caps
+ 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
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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
-2004-12-19 Edward Hervey <bilboed@bilboed.com>
+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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_change_state):
- Event handling (fixes #159986).
+2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add BLZ0 (Blizzard's version of DivX) fourcc.
+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>
-2004-12-18 David Schleef <ds@schleef.org>
+ * 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).
- * gst/tta/gstttadec.c: (gst_tta_dec_link): And yet another
- portability fix.
+2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-18 David Schleef <ds@schleef.org>
+ 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.
- * gst/tta/ttadec.h: Disable some header code that isn't used
- and clearly isn't portable.
+2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-18 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * 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.)
+=== release 0.10.6 ===
-2004-12-19 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * 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.
+2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ dist more win32 files
+ Original commit message from CVS:
+ dist more win32 files
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_sint):
- That was very stupid.
+2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix possible crasher.
+2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_uint),
- (gst_matroska_ebmlnum_sint), (gst_matroska_demux_parse_blockgroup):
- Lace sizes can be zero.
+2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * configure.ac:
+ * win32/common/config.h:
+ new prerelease
+ Original commit message from CVS:
+ new prerelease
- * ext/musepack/gstmusepackdec.cpp:
- Fetch error return values. Fixes #161624.
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
- Really EOS.
+2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
- Work for truncated (unfinished download etc.) files. Fixes #160514.
+2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Fix for integer overflow. Makes #156001 not crash. Probably masks
- the real bug.
+2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
- * gst/ac3parse/gstac3parse.c: (plugin_init):
- Parsers never have ranks. Fixes #159651.
+2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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.
- * 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
+2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-16 David Schleef <ds@schleef.org>
+ 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.
- * configure.ac: look for musepack headers as musepack/*.h
- (fixes #159847)
- * ext/musepack/gstmusepackdec.h: use <musepack/*.h>
- * ext/musepack/gstmusepackreader.h: same
+2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_auds_with_data):
- Read extradata correctly (fixes #155879).
+2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 David Schleef <ds@schleef.org>
+ * 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
- * gst/audioscale/gstaudioscale.c: allow passthru of >2 channel
- audio. does _not_ attempt or allow conversion unless channels
- is 1 or 2.
+2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-12-16 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ 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
- * tools/gst-launch-ext-m.m: fix mpeg and vob pipelines
+2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ * 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:
+ update libtool versioning
+ Original commit message from CVS:
+ update libtool versioning
- * gst/audioscale/gstaudioscale.c: the resample library only
- handles 1 or 2 channels. Change caps to compensate.
+2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * configure.ac:
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
- * 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...
+2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add h264.
+2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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.
- * gst-libs/gst/audio/Makefile.am:
- Try to fix buildbot.
+2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ Mention bug #336617 closed by recent commit
+ Original commit message from CVS:
+ Mention bug #336617 closed by recent commit
- * gst/tcp/gstmultifdsink.c:
- Clean up and uniformize debugging.
+2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
-2004-12-16 Edward Hervey <bilboed@bilboed.com>
+ 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.
- Reviewed by: Ronald S. 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
- * 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.
+2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_pad_link):
- Fix memleak (#154815).
+ 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
-2004-12-16 James Bowes <bowes@cs.dal.ca>
+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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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).
+ 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.
-2004-12-16 Christophe Fergeau <teuf@gnome.org>
+2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/playback/gstdecodebin.c: (new_pad), (close_link):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Fix memleaks (#157233).
+2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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!
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
- * gst-libs/gst/resample/resample.c: (gst_resample_close):
- * gst-libs/gst/resample/resample.h:
- * gst/audioscale/gstaudioscale.c:
- Fix memleak (#159215).
+ 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.
-2004-12-16 Toni Willberg <toniw@iki.fi>
+2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-15 David Schleef <ds@schleef.org>
+ 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.
- * 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.
+2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
-2004-12-15 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+ 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.
- Reviewed by: David Schleef <ds@schleef.org>
+2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
- * 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):
+ 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.
-2004-12-14 David Schleef <ds@schleef.org>
+2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
- * gst/festival/gstfestival.c: (gst_festival_chain): Set the
- output rate to 16000. Should fix #160235.
+ 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.
-2004-12-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Add typefinding for mpeg2 pes streams
+ 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.
-2004-12-13 David Schleef <ds@schleef.org>
+2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * common:
+ * configure.ac:
+ 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
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/riff/riff-media.c:
- forgot to add h2.64 to avidemux template caps
+ * 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
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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 :(
+ 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)
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-2004-12-12 Christian Fredrik Kalager Schaller <christian at fluendo dot com>
+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>
- * configure.ac: Update polyaudio requirement to 0.7
- * ext/polyp/polypsink.c: (create_stream): add patch from iain (158258)
+ * 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
-2004-12-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/interleave/deinterleave.c:
- fix my name's spelling! :)
+ * 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
-2004-12-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
- * 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
+ 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
- put the same mail address for Zaheer Abbas Merali everywhere
+2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (get_relative):
- Check for non-NULL before accessing member (end-of-chain).
+2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
-2004-12-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * configure.ac: move GCONF macro outside conditional for the am
- conditional. Fixes #160439
+2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-08 David Schleef <ds@schleef.org>
+ 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).
- * tools/gst-visualise-m.m: Switch to elements that currently
- exist.
+2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- We love wrong commas.
+2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_src_query):
- Don't set DEFAULT, unsupported - makes length display incorrectly
- in some cases.
+2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ * 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
- * 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
+2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ 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
- * gst/monoscope/README: Update information to be more correct
- * gst/monoscope/convolve.c: Relicense to LGPL
- * gst/monoscope/convolve.h: Relicense to LGPL
+2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-06 Arwed v. Merkatz <v.merkatz@gmx.net>
+ 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
- * 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
+2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * 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
- * 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)
+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).
-2004-12-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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).
- * 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.
+2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/apetag/apedemux.c: (gst_ape_demux_parse_tags),
- (gst_ape_demux_stream_init):
- Forward tags, too.
+2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_init):
- Let's make sure we're done typefinding when detecting tags.
+2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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)
- * 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.
+2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-03 Ronald S. 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).
- * ext/musepack/gstmusepackdec.cpp:
- There's also floating point libmusepacks.
+2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix logic bug.
+2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Yay, another one.
+2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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).
+2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
-2004-12-01 Christophe Fergeau <teuf@gnome.org>
+ 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).
- * ext/mad/gstmad.c: (mpg123_parse_xing_header): fix xing header
- detection on mono and stereo mp3 files.
+2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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).
+2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-01 Ronald S. 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/faad/gstfaad.c: (gst_faad_srcconnect), (gst_faad_chain):
- Oops, remove debug.
+2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
+ configure.ac: back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ back to CVS.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/law/alaw-decode.c: (alawdec_getcaps):
- * gst/law/mulaw-decode.c: (mulawdec_getcaps):
- Prevent warnings when negotiating caps (fixes #159338).
-
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- Remove old leftover that shouldn't be there...
-
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
- Don't forward DISCONT events (fixes #159684).
-
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Actually leave the loop if we failed to sync. Don't crash.
-
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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. :-(.
-
-2004-11-28 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
- more overwriting protection due to modifying channels one by one
- instead of all at once
-
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_normalize):
- Normalize using absolute values.
+=== release 0.10.5 ===
-2004-11-28 Julien MOUTTE <julien@moutte.net>
+2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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>
-
- * 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.
-
-2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
+ * 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
- * 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.
+2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-27 Christophe Fergeau <teuf@gnome.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
- * 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
+2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Don't forget bass if it's there. Else left channel is silent...
+2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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().
- * ext/a52dec/gsta52dec.c: (gst_a52dec_channels),
- (gst_a52dec_change_state):
- Advertise that we can do surround sound.
+2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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!).
+2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Christophe Fergeau <teuf@gnome.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).
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
- call parent dispose method
+2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
+ configure.ac: back to trunk
+ Original commit message from CVS:
+ * configure.ac:
+ back to trunk
- * 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.
+=== release 0.10.4 ===
-2004-11-27 Arwed v. Merkatz <v.merkatz@gmx.net>
+2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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.
-
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/theora/theoradec.c: (theora_dec_chain):
- Fix ilog mask range overflow.
-
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-11-26 Christophe Fergeau <teuf@gnome.org>
-
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element): Removed 2 obsolete comments
-
-2004-11-26 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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)
-
-2004-11-25 Christophe Fergeau <teuf@gnome.org>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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/playback/gstdecodebin.c: (gst_decode_bin_factory_filter):
- We sometimes need parsers for playback, so add those too.
+2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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>
+2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * 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>
+ 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)
- * configure.ac:
- back to CVS
+2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
-=== release 0.8.6 ===
+ 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.
-2004-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * NEWS:
- * RELEASE:
* configure.ac:
+ * docs/libs/tmpl/gstringbuffer.sgml:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
- releasing 0.8.6, "IOU Love"
+ * 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).
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/playback/gstplaybasebin.c:
- Fix unplayable files error handling. Fixes #158365
+2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c:
- Fix broken mp3 typefinding. Fixes #158375
+ 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).
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/ogg/gstoggdemux.c:
- Fix sync on broken files. Fixes #158976
+2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- patch by: Edward Hervey <bilboed@bilboed.com>
+2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
- * ext/libpng/gstpngenc.c:
- Copy over buffer properties. Fixes #158832
+ 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.
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * ext/dvdread/dvdreadsrc.c:
- Fixes invalid reads (#158462)
+ 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).
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- Probe less and cache it. Fixes #159187.
+ 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().
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-07 13:01:21 +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/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().
- * configure.ac:
- Fix mpeg2enc configure check (similar to mplex check below).
+2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
-2004-11-14 Koop Mast <kwm@rainbow-runner.nl>
+ 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).
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- Fix for gcc-2.95 (fixes #158221).
+ 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
-2004-11-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- Re-add clock distribution hack (until new core is released).
- Fixes #158125.
+ 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).
-2004-11-13 Arwed v. Merkatz <v.merkatz@gmx.net>
- * configure.ac:
- fix mplex configure check segfaulting on some systems (bug #140994)
+2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-13 Benjamin Otte <otte@gnome.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.
- * 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
+2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
-2004-11-13 Jan Schmidt <thaytan@mad.scientist.com>
+ 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).
- * 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>
+2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/volume/gstvolume.c:
- Allow buffer-frames=0.
+ 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.
-2004-11-12 Iain <iaingnome@gmail.com>
+2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: Check for polypaudio
+ * ext/ogg/gstoggmux.c:
+ changed more than 5 lines
+ Original commit message from CVS:
+ changed more than 5 lines
- * ext/Makefile.am: Build the polyp dir
+2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/polyp: The polypsink sources.
+ 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
-2004-10-30 Iain <iaingnome@gmail.com>
+2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
- * gst/interleave/interleave.c (interleave_unlink): Change the src pads
- caps to reflect the new number of channels.
+ 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)
-2004-11-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2004-11-11 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- No warnings (#157986).
+2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Prefer apev1/2 and id3v1 (at end of file) over musepack.
+2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream):
- Signal no-more-pads (so it works in playbin).
+2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ext/ogg/README:
+ * ext/ogg/gstoggmux.c:
+ debug updates
+ Original commit message from CVS:
+ debug updates
- * ext/musepack/gstmusepackreader.cpp:
- Workaround for older core.
+2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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).
+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>
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/videorate/Makefile.am:
+ fix wim's commit
+ Original commit message from CVS:
+ fix wim's commit
- * sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * 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>
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+ * ext/ogg/gstoggmux.c:
+ clean up debug messages
+ Original commit message from CVS:
+ clean up debug messages
-2004-11-11 Jan Schmidt <thaytan@mad.scientist.com>
+2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ * 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
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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).
+ * 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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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).
+ 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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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)
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Don't assert (#157853).
+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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
- * 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).
+ 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).
-2004-11-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- Disable halfway-seek for pending release (since it needs a new
- core release).
+ 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>)
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- add autoprobe-fps property so we can separate autoprobing parts
+ 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.
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
-
-2004-11-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Submitted by: Luca Ferretti <elle.uca@infinito.it>
+ 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.
- * po/LINGUAS:
- * po/it.po:
- Add Italian
+2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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/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..
+2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ * 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
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- We can remove this hack now.
+2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Actually always send a discont (cornercase when resending the
- same serial-tagged chain twice).
+2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+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).
-2004-11-08 Wim Taymans <wim@fluendo.com>
+2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
- Don't segfault on NULL data.
+ 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.
-2004-11-08 Wim Taymans <wim@fluendo.com>
+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>
- * 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.
+ * 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).
-2004-11-07 David Schleef <ds@schleef.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).
- * testsuite/gst-lint: Check for pad templates that aren't statically
- scoped.
+2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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().
+2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_loop):
- Fix interrupt event handling (#144436).
+2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_typefind):
- Hide unused glory.
+2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+ 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).
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisenc.c: (raw_caps_factory):
- Fix weird caps (#157548).
+ 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.
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/rtp/gstrtpgsmparse.c: (gst_rtpgsm_caps_nego):
- Add missing NULL terminator (#157543).
+2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/videotestsrc/videotestsrc.c: (paint_setup_Y41B),
- (paint_hline_Y41B), (paint_setup_Y42B), (paint_hline_Y42B):
- Added two more colorspaces.
+2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ 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
- * 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.
+2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ 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
- * 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.
+2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-05 Benjamin Otte <otte@gnome.org>
+ 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.
- * ext/mad/gstmad.c: (gst_mad_chain):
- don't overflow data buffer. Flush not needed sync data when syncing
- failed.
+2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-04 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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
-2004-11-03 Wim Taymans <wim@fluendo.com>
+2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
- * 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.
+ * ChangeLog:
+ changelog foo
+ Original commit message from CVS:
+ changelog foo
-2004-11-03 Christophe Fergeau <teuf@gnome.org>
+2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- * gst/videoscale/videoscale.c: (videoscale_find_by_structure):
- leak fixes
+ * 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
-2004-11-03 Wim Taymans <wim@fluendo.com>
+=== release 0.10.3 ===
- * 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>
+2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggdemux.c:
- * gst/tags/gstvorbistag.c:
- Plug leaks.
+ * 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 0.10.3
+ Original commit message from CVS:
+ Releasing 0.10.3
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- lotsa memleaks today. But they're all small...
+ 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)
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- another memleak crushed
+ 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).
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- fix memleak
+ 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.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Hack to prevent crash when going to READY inside signal handler
- while this function is active.
+ * 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
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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/tcp/gstmultifdsink.c:
+ doc fixes
+ Original commit message from CVS:
+ doc fixes
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/Makefile.am:
- We need the marshallers for decodebin, too.
+ 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
-2004-10-30 David Schleef <ds@schleef.org>
+2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find): Make
- quicktime typefinding work with 64-bit offsets.
+ 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.
-2004-10-30 Jan Schmidt <thaytan@mad.scientist.com>
+2006-02-07 17:44:53 +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.
+ 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.
-2004-10-29 Martin Pitt <martin.pitt@canonical.com>
+2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
- Don't touch buffer if it is of size 0 (fixes #151064).
+2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/ogg/gstoggdemux.c: (gst_ogg_demux_push), (gst_ogg_pad_push):
- Synchronized discont handling.
+2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_push):
- Make seeking sort-of exact again (fixes #156387).
+2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format):
- Really don't touch read-only buffers (#156563).
+2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
-2004-10-29 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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).
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak (#155223).
+ 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).
-2004-10-29 Wim Taymans <wim@fluendo.com>
+2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
- * 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.
+ 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.
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (thread_error), (setup_source),
- (gst_play_base_bin_change_state):
- Improve error reporting.
+ 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.
-2004-10-28 Wim Taymans <wim@fluendo.com>
+2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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>
+ 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.
- * po/LINGUAS:
- * po/nb.po:
- Added Norwegian Bokmaal translation
+2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-27 Ronald S. Bultje <rbultje@ronald.bitfreak.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"
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- Don't break on options (fixes #156488).
+2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- * ext/cdaudio/Makefile.am:
- * sys/Makefile.am:
- fix build on older automake
+2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-26 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-26 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * gst/level/gstlevel.c:
- synchonised naming of pads and pad-templates
+2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
-2004-10-26 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-26 Iain <iaingnome@gmail.com>
+ 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.
- * 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.
+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.
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_init),
- (gst_gnomevfssink_handle_event), (gst_gnomevfssink_chain):
- Implement seeking.
+ 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.
-2004-10-25 James Henstridge <james@jamesh.id.au>
+2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
- Reviewed by: David Schleef <ds@schleef.org>
+ 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.
- * 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.
+2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
-2004-10-25 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- reviewed by: Ronald Bultje <rbultje at gnome dot org>
+2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix for some v4l cards which hang in v4lsrc
+ * configure.ac:
+ Improve error message for liboil missingness.
+ Original commit message from CVS:
+ Improve error message for liboil missingness.
-2004-10-25 Wim Taymans <wim@fluendo.com>
+2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2004-10-25 Wim Taymans <wim@fluendo.com>
+2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * 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.
+ * 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.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2004-10-25 Takao Fujiwara <Takao.Fujiwara@Sun.COM>
+2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * tests/check/pipelines/theoraenc.c:
+ comment on my understanding
+ Original commit message from CVS:
+ comment on my understanding
- * 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).
+2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * common:
+ * tests/check/pipelines/theoraenc.c:
+ reformat to fit 80 chars
+ Original commit message from CVS:
+ reformat to fit 80 chars
- * ext/theora/theoradec.c: (theora_dec_chain):
- Export bitrate as metadata.
+2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/speex/gstspeexdec.c: (gst_speex_dec_init),
- (speex_dec_convert):
- sinkconvert function so oggdemux can get the file length (totem).
+2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-25 James Morrison <ja2morri@csclub.uwaterloo.ca>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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.
+2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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?
+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
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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>
- * 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.
+ * gst-plugins-base.spec.in:
+ update with latest files
+ Original commit message from CVS:
+ update with latest files
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
- * gst/asfdemux/gstasfmux.c:
- * gst/avi/gstavimux.c:
- Framerate.
+ 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
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_set_property):
- Fix properties (channel, norm, frequency).
+ 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
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_get_property):
- Flag typo.
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_set_defaults):
- No warnings.
+ 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.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_clear_format_list):
- Fix hang.
+ * 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
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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)
- * 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.
+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>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Quiet.
+2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ generate win32/common/config.h
+ Original commit message from CVS:
+ generate win32/common/config.h
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Don't mix bytes and samples.
+2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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>
+2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/Makefile.am:
- Fix the build fixes.
+ 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.
-2004-10-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2004-10-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- We love it if files that start at zero work too...
+ 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.
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate):
- Handle files with missing EOS headers.
+ 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
-2004-10-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tcp/gsttcpserversink.c:
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_init_send):
- Zero some variables first (need for accept not to return EINVAL)
+ 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.
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
-
-2004-10-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- * ext/Makefile.am:
- * ext/raw1394/Makefile.am:
- fix the build
+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
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
- * 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.
+ 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.
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event):
- Silence.
+ 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.
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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.
+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.
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
- * 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.
+ 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.
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.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.
+ * autogen.sh:
+ * common:
+ up automake requirement to 1.7
+ Original commit message from CVS:
+ up automake requirement to 1.7
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- I swear, this is the last time I touch this.
+ gst/playback/gstplaybasebin.c: Free iterator when done.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Free iterator when done.
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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-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
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
- Added cdaudio to wrong list.
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
+
+=== release 0.10.2 ===
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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
+ * 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
-2004-10-13 Wim Taymans <wim@fluendo.com>
+2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
- Decoding the header first fixes some problems in resyncing
- in more mp3s.
+ 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.
-2004-10-12 Wim Taymans <wim@fluendo.com>
+2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ 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)
-2004-10-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_open),
- (cdparanoia_event), (cdparanoia_query):
- Get rid of hideous lead-in.
+ 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
-2004-10-11 Wim Taymans <wim@fluendo.com>
+2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Wrong var used to get g_list_next.
+ * 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
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_get), (cdparanoia_open):
- Report discid as metadata, add duration.
+ 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
-2004-10-11 Wim Taymans <wim@fluendo.com>
+2006-01-14 12:52:22 +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.
+ 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.
-2004-10-11 Benjamin Otte <otte@gnome.org>
+2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_src_link), (gst_v4lsrc_getcaps):
- Fix for webcams that support only specific width or height
+ * gst-plugins-base.spec.in:
+ remove version suffix
+ Original commit message from CVS:
+ remove version suffix
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. 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.
- * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
- Fix wrong discont event setup (fixes #154967).
+2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-09 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Error out on invalid data (fixes #154807).
+ 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.
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst-plugins-base.spec.in:
+ update spec.in file
+ Original commit message from CVS:
+ update spec.in file
- * ext/dvdread/dvdreadsrc.c: (_read):
- Make titles > 0 work again (fixes #154834).
+2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.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-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_template_caps):
- WMV3 missing in template caps.
+2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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.
+2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-10-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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"...
+2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ 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)
- * 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.
+2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
-2004-10-08 Milosz Derezynski <internalerror.rez@fhtw-berlin.de>
+ 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.
- * ext/cdparanoia/gstcdparanoia.c: (_do_init),
- Added reworked patch from #154903 from milosz derezynski (deadchip).
+2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-08 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-08 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-10-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/Makefile.am:
+ Original commit message from CVS:
+ * gst-libs/gst/Makefile.am:
- * 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.
+2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-07 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
-2004-10-07 Wim Taymans <wim@fluendo.com>
+ 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
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find),
- (plugin_init):
- Added lame audio/x-ac3 typefind function.
+2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ 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.
- * configure.ac:
- bump nano to cvs
+2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.8.5 ===
+ 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).
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.5, "Take You On"
+ 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
-2004-10-06 Wim Taymans <wim@fluendo.com>
+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/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, ...).
+2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot 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:
- bump for prerelease
+2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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/wavparse/gstwavparse.c:
- add ATRAC3 to STATIC CAPS to fix a warning
+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).
- * gst/matroska/ebml-read.c:
- * gst-libs/gst/riff/riff-read.c:
- fix typos
+2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ tests/check/Makefile.am: Fix test case compilation.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix test case compilation.
- * gst-libs/gst/riff/riff-media.c:
- generate caps for ATRAC3 audio streams
+2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/realmedia/rmdemux.c:
- generate caps for ATRAC3 audio streams
+ 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).
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/Makefile.am
- * gst/wavparse/riff.h
- * gst/wavparse/wavparse.vcproj
- riff.h removal (unused and duplication with riff-ids.h)
+ 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.
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.h
- remove duplicated defines for audio codec codes
+ 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-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)
+2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_loop):
- Actually _do_ negotiation. Pass gdouble as arg instead
- of guint64 for the framerate.
+2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+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.
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- written by: Gora Mohanty <gora_mohanty@yahoo.co.in>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * po/LINGUAS:
- * po/or.po:
- add Oriya translation
+=== release 0.10.1 ===
-2004-10-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- Prevent overwrite of size member. Makes audio sound crappy.
+ * 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
-2004-10-05 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add rmvb to the list of known RealMedia extensions
+ * ChangeLog:
+ * gst/typefind/gsttypefindfunctions.c:
+ iLBC30 and iLBC20 added to typefind.
+ Original commit message from CVS:
+ iLBC30 and iLBC20 added to typefind.
-2004-10-05 Wim Taymans <wim@fluendo.com>
+2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * 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
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- There was something wrong with the index massaging.
+ * 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
-2004-10-04 Wim Taymans <wim@fluendo.com>
+2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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/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
-2004-10-04 Ronald S. 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/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.
+2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
-2004-10-04 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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>
+2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- * configure.ac:
- you need at least 1.0.4 of speex
+ 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)
-2004-10-04 Iain <iaingnome@gmail.com>
+2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- * ext/speex/gstspeexdec.h: Revert the includes changes.
-
- * ext/speex/gstspeexenc.[ch]: Revert the includes changes.
-
-2004-09-30 Iain <iaingnome@gmail.com>
-
- * sys/sunaudio/gstsunaudio.c (gst_sunaudiosink_open): Use the device
- found during init or set as a property instead of hardcoding /dev/audio
-
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-10-03 Christophe Fergeau <teuf@gnome.org>
-
- * gst/realmedia/rmdemux.c: (gst_rmdemux_parse_mdpr):
- fixed compilation
-
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-10-01 Christian Schaller <christian@fluendo.com>
-
- * ext/ivorbis/vorbis.c:
- Patch from Phil Blundell (Bug 152341)
-
-2004-10-01 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-10-01 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-10-01 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2004-10-01 Ronald S. 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)
- * ext/dvdnav/dvdnavsrc.c: (dvdnav_handle_navigation_event):
- Don't use g_print(); use GST_DEBUG().
+2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-01 Ronald S. 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
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads):
- Handle EOS properly.
+2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
- * 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/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).
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Throw error if we didn't recognize the stream. Fixes #152289.
+2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- Fix negotiation.
+2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Francis Labonte <francis_labonte@hotmail.com>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak.
+ 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.
-2004-10-01 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+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).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
- * sys/sunaudio/gstsunaudio.c: (gst_sunaudiosink_setparams):
- Solve #152805.
- * sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_mute):
- Solve 152806.
+ * ChangeLog:
+ * ext/Makefile.am:
+ * gst-plugins-base.spec.in:
+ updates to activate cdparanoia plugin
+ Original commit message from CVS:
+ updates to activate cdparanoia plugin
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
- * 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>
+ 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.
- * 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.
+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.
-2004-09-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.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
+ 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.
-2004-09-26 Iain <iaingnome@gmail.com>
+2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2004-09-26 Iain <iaingnome@gmail.com>
+2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
- * 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.
+ 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.
-2004-09-25 Christian Schaller <christian@fluendo.com>
+2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Update SPEC file to be usable in conjunction with Fedora Core,
- Fedora.us and freshrpms packages
- * Fix typo in multifilesrc test Makefile
+ 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.
-2004-09-24 Wim Taymans <wim@fluendo.com>
+2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Only signal the no_more_pads signal when we have
- added the stream to our list.
+ 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.
-2004-09-24 Wim Taymans <wim@fluendo.com>
+2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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):
+ * 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:
- 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):
+ * 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/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.
-
-2004-09-16 Iain <iaingnome@gmail.com>
-
- * gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int): Free the
- try caps.
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_data):
- Just hardcode for raw audio then. AVI audio sucks.
-
-2004-09-15 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * 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.
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_data):
- Try to fix a/v sync issues.
-
-2004-09-15 David Schleef <ds@schleef.org>
-
- * 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
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.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?
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Volume is a double not a float.
-
-2004-09-15 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-09-15 Wim Taymans <wim@fluendo.com>
-
- * 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).
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.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).
-
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2004-09-14 Brian Cameron <brian.cameron@sun.com
-
- * 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.
-
-2004-09-17 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-09-17 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
- (qtdemux_parse_trak):
- Don't crash by dividing by zero (see sample movie in #126922).
-
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Don't touch non-existing data (fixes crash on file in #140147).
-
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c:
- (gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
- Handle double disposals, and proper change of URIs.
-
-2004-09-13 Martin Eikermann <meiker@upb.de>
-
- * gst/mpegstream/gstmpegparse.c:
- fix synchronistation for streams recorded from digital PCR
- fixes bug #119376
-
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * 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>
-
- * configure.ac:
- bump nano to cvs
-
-=== release 0.8.4 ===
-
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: releasing 0.8.4, "Alias"
-
-2004-08-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/theora/Makefile.am:
- fix makefile. Fixes #151462.
-
-2004-08-30 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: David Schleef
+ * 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
- * configure.ac:
- * sys/Makefile.am:
- rename our detection macro for V4L2. Fixes #151236.
+2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- Patch by: David Schleef
+2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* configure.ac:
- check to define LAMEPRESET. Fixes #151232.
-
-2004-08-27 David Schleef <ds@schleef.org>
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * 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.
+=== release 0.10.0 ===
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- bump nano for prerelease
-
-2004-08-27 David Schleef <ds@schleef.org>
-
- * sys/sunaudio/Makefile.am: Add sunaudiosrc patch from Bala
- * sys/sunaudio/gstsunaudiosrc.c:
- * sys/sunaudio/gstsunaudiosrc.h:
-
-2004-08-27 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- * 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
-
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Zaheer Abbas Merali
-
- * 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>
-
- * configure.ac:
- Clean up the test for lame presets
-
-2004-08-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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.
-
-2004-08-18 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-18 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-17 David Schleef <ds@schleef.org>
-
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: Fix caps to handle full range
- of rates and channels. Make debugging less obnoxious.
-
- 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:
-
-2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/gstaudioscale.h:
- made audioscale resample from any sample rate to any sample rate
-
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/libpng/gstpngdec.c:
- error out on unsupported types
-
-2004-08-17 Iain <iaingnome@gmail.com>
-
- * ext/flac/gstflacenc.c (gst_flacenc_update_quality): Only set the
- mid_side and loose_mid_side properties if its a stereo stream.
-
-2004-08-17 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):
- Add a debug line.
-
-2004-08-17 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-17 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Wim Taymans
-
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstmultifdsink.c:
- fix index comparison, should include 0
-
-2004-08-16 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2004-08-16 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-08-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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
-
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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
-
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_get_property):
- whoops forgot break, thanks teuf
-
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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
-
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_init):
- added getcaps function so samplerate doesnt get fixated to silly values
-
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_src_link):
- revert previous fix
-
-2004-08-12 Johan Dahlin <johan@gnome.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.
-
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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
-
-2004-08-12 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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
-
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init):
- set default compression ratio parameter to 0.0 so bitrate parameter
- works :)
-
-2004-08-11 David Schleef <ds@schleef.org>
-
- * gst/tcp/gstfdset.c: Fix compile problem on OS/X.
-
-2004-08-11 David Schleef <ds@schleef.org>
-
- * gst/mpeg1sys/gstmpeg1systemencode.c: Oops, this was correct
- before.
-
-2004-08-11 David Schleef <ds@schleef.org>
-
- * gst-libs/gst/video/videosink.h: Change copyright block to LGPL.
-
-2004-08-11 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-08-11 Wim Taymans <wim@fluendo.com>
-
- * 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).
-
-2004-08-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * po/LINGUAS:
- * po/sq.po:
- Added Albanian translation (Laurent Dhima)
- * po/cs.po:
- updated
-
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/lame/gstlame.c:
- fix/add debugging
-
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- assign all TOO_LAZY's to a real category. Thanks to Warthy Warthog.
-
-2004-08-06 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):
- Make sure we don't try to read more from a client that what
- ioctl says us or we deadlock.
-
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/nl.po:
- * po/sv.po:
- updated translations
-
-2004-08-04 Benjamin Otte <otte@gnome.org>
-
- * 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>
-
- * 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
-
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/uk.po:
- updated translation
-
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
- add debugging for display PAR calculation
-
-2004-08-02 David Schleef <ds@schleef.org>
-
- * configure.ac: Fix mikmod CFLAGS.
-
-2004-07-27 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-08-02 Iain <iain@prettypeople.org>
-
- * 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.
-
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: bump nano back to development
-
-=== release 0.8.3 ===
-
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: releasing 0.8.3, "Water"
-
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch from: Benjamin Otte
-
- * ext/lame/gstlame.c: (gst_lame_src_link), (gst_lame_init):
- add link function to lame. Fixes #148986.
-
-2004-08-02 Johan Dahlin <johan@gnome.org>
-
- * gst/multipart/multipartmux.c (gst_multipart_mux_next_buffer):
- fix debugging log
-
-2004-07-30 David Schleef <ds@schleef.org>
-
- * gst/videomixer/Makefile.am: Fix things that should have been
- fixed in the last checkin.
-
-2004-07-30 David Schleef <ds@schleef.org>
-
- * gst/multipart/Makefile.am: Fix things that should have been
- fixed in the last checkin.
-
-2004-07-30 David Schleef <ds@schleef.org>
-
- * testsuite/multifilesink/Makefile.am: Fix unused variable.
-
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-30 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
- fix default for newmedia flag
-
-2004-07-30 Wim Taymans <wim@fluendo.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.
-
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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
-
-2004-07-30 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-29 David Schleef <ds@schleef.org>
-
- * configure.ac: make test for audiofile more strict
-
-2004-07-25 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- give different names to typefind functions
-
-2004-07-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-27 Steve Lhomme <steve.lhomme@free.fr>
-
- * win32/gst.sln:
- * gst-libs/gst/*/*.vcproj:
- * gst/*/*.vcproj:
- more working plugins
-
-2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * testsuite/alsa/Makefile.am:
- * testsuite/alsa/srcstate.c:
- add test for alsasrc changing state
-
-2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst/silence/gstsilence.c: (gst_silence_init), (gst_silence_link),
- (gst_silence_get):
- * gst/silence/gstsilence.h:
- fix silence generation for 16bit raw audio
-
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot 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.
-
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot 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.
-
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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-libs/gst/audio/gstaudiofilter.c:
- Clean the local include
-
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
-
- * 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
-
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
-
- * gst/matroska/ebml-write.c:
- Fix some 64 bits constants to be glib friendly
-
-2004-07-24 Steve Lhomme <steve.lhomme@free.fr>
-
- * 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
-
-2004-07-23 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- don't enfore negotiation from source side, it breaks
- sinesrc ! audioconvert ! osssink
-
-2004-07-22 David Schleef <ds@schleef.org>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add typefind
- for ELF files, since they can easily be recognized as audio/mpeg.
- (bug #147441)
-
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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.
-
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videoscale/videoscale.c: (gst_videoscale_planar411),
- (gst_videoscale_scale_nearest_16bit):
- comment algorithm
-
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- no point in doing any chaining if the pad we want to push from
- isn't usable.
-
-2004-07-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Fix double end-to-native symbol conversion (#148021).
-
-2004-07-20 David Schleef <ds@schleef.org>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate):
- Don't use an Atom that doesn't exist.
-
-2004-07-20 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_write),
- (gst_multifdsink_queue_buffer):
- * gst/tcp/gstmultifdsink.h:
- More multifdsink stats. Avoid deadlock by releasing locks
- before sending out a signal.
-
-2004-07-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS:
- * po/hu.po:
- added Hungarian translation (Laszlo Dvornik)
-
-2004-07-20 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-20 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-20 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_write):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp-marshal.list:
- Added get_stats method that returns a GValueArray of
- stats values.
-
-2004-07-19 Benjamin Otte <otte@gnome.org>
-
- * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
- make sure longname, description and author are valid UTF-8
-
-2004-07-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-16 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-07-15 Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-07-15 Benjamin Otte <otte@gnome.org>
-
- * 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.
-
-2004-07-12 Benjamin Otte <otte@gnome.org>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_alloc_buffer):
- use bufferpools
-
-2004-07-14 Thomas Vander Stichele <thomas at apestaart dot 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.
-
-2004-07-13 Johan Dahlin <johan@gnome.org>
-
- * gst/playback/gstplaybin.c (gst_play_bin_set_property)
- (gst_play_bin_get_property): Impl.
-
-2004-07-13 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-13 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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
-
-2004-07-08 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/Makefile.am:
- Add headers to noinst
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * tools/gst-launch-ext-m.m:
- * tools/gst-launch-ext.1.in:
- convert to the third millenium
-
-2004-07-07 David Schleef <ds@schleef.org>
-
- * sys/dxr3/Makefile.am: noinst_SOURCES should be nodist_SOURCES
-
-2004-07-07 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-07 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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.
-
-2004-07-07 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-07 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-07-07 Wim Taymans <wim@fluendo.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.
-
-2004-07-06 David Schleef <ds@schleef.org>
-
- * gst/playback/Makefile.am: 'test' in bin_PROGRAMS? Are you
- serious? (Fixed, obviously.)
-
-2004-07-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * 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>
-
- * 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.
-
-2004-07-02 Benjamin Otte <otte@gnome.org>
-
- * 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.
-
-2004-06-29 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2004-06-26 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_chain),
- (plugin_init):
- fix debugging. add category.
-
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- fix wrong offsets
-
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-25 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- better debug, don't override OFFSET and OFFSET_END
-
-2004-06-25 Iain <iain@prettypeople.org>
-
- * gst-libs/gst/media-info/media-info-priv.c (gmi_set_mime): Add
- name=source for the wavparse pipeline.
-
-2004-06-24 Johan Dahlin <johan@gnome.org>
-
- * 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.
-
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisenc.c: (raw_caps_factory),
- (gst_vorbisenc_setup), (gst_vorbisenc_set_property):
- respect minimum bitrate; same could be done for max bitrate
-
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisenc.c: (raw_caps_factory),
- (gst_vorbisenc_setup):
- fix sample rate range
-
-2004-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-24 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-24 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-24 Benjamin Otte <otte@gnome.org>
-
- * ext/kio/Makefile.am:
- fix for builddir != srcdir and distcheck
-
-2004-06-24 Benjamin Otte <otte@gnome.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:
- * 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.
-
-2004-06-23 David I. Lehn <dlehn@users.sourceforge.net>
-
- * ext/Makefile.am: s/DTS_DIR=dvdread/DTS_DIR=dts/
-
-2004-06-23 Colin Walters <walters@redhat.com>
-
- * m4/Makefile.am: Distribute gst-fionread.m4.
-
-2004-06-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: back to dev
-
-2004-06-23 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2004-06-23 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2004-06-22 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-22 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-22 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_init):
- Initialize the property as well.
-
-2004-06-22 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * sys/osxvideo/Makefile.am:
- Workaround so that the osxvideo .so file gets linked with the
- Cocoa, OpenGL and QuickTime frameworks
-
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * sys/osxaudio/Makefile.am:
- Workaround so that the osxaudio .so file gets linked with the
- CoreAudio framework
-
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * configure.ac:
- Whoops, my fault...fixed build issues
-
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot 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.
-
-2004-06-18 Wim Taymans <wim@fluendo.com>
-
- * ext/sidplay/gstsiddec.cc:
- Fix negotiation and set correct end offset.
-
-2004-06-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: branch and prerelease
-
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2004-06-17 Wim Taymans <wim@fluendo.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/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.
-
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- fixed a potential leak with previous commit
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
-
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
- Added missing refcount, fixes bug #144425
- Cheers Tim for finding the bug
-
-2004-06-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-14 David Schleef <ds@schleef.org>
-
- * 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:
-
-2004-06-14 Benjamin Otte <otte@gnome.org>
-
- * 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.
-
-2004-06-14 David Schleef <ds@schleef.org>
-
- * gst/volume/demo.c: (value_changed_callback): exp10() is not
- standard. Thank you for playing.
-
-2004-06-14 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/volume/.cvsignore:
- * gst/volume/Makefile.am:
- * gst/volume/demo.c: (value_changed_callback), (idler),
- (setup_gui), (main):
- added small demo app
-
-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.
-
-2004-06-12 Christophe Fergeau <teuf@gnome.org>
-
- * gst/tags/gstvorbistag.c: replaced a g_warning which I added in my
- previous commit with GST_DEBUG
-
-2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- add a header check for a dvdread header in dvdnav. Fixes #133002
-
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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)
-
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * 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>
-
- * 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>
-
- * 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>
-
- * 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>
-
- * configure.ac:
- * ext/kio/Makefile.am:
- check for qt's moc preprocessor explicitly and use it
-
-2004-06-03 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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)
-
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_get_type):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- check error condition on available samples correctly
-
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-06-02 Wim Taymans <wim@fluendo.com>
-
- reviewed by: Johan
-
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_loop):
- Oops
-
-2004-06-02 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-06-02 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversrc.c:
- - portability fix, to compile on OSX
- (fixes #143146)
-
- * sys/osxaudio/gstosxaudioelement.c:
- * sys/osxaudio/gstosxaudiosink.c:
- * sys/osxaudio/gstosxaudiosrc.c:
- - compilation warnings on OSX
- (fixes #143153)
-
-2004-06-02 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/vorbis/vorbisdec.c : sign warning fixes
-
- * gst-libs/gst/mixer/mixertrack.c :
- do no use defines which are glib 2.4 specific
-
-2004-06-01 Christophe Fergeau <teuf@gnome.org>
-
- * 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)
-
-
-2004-06-01 Owen Fraser-Green <owen@discobabe.net>
-
- * 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
-
-
-2004-06-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio):
- improve error messages on open
-
-
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- check if v4l-conf is in path
-
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/media-info/media-info-priv.c: (gmi_set_mime):
- change assert to a more readable error message
-
-2004-05-31 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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)
-
-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.
-
-2004-05-28 Wim Taymans <wim@fluendo.com>
-
- * 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>
-
- * 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>
-
- * 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
-
-2004-05-25 David Schleef <ds@schleef.org>
-
- * gst/ffmpegcolorspace/mem.c: malloc() is in stdlib.h, not malloc.h
-
-2004-05-24 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-05-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/Makefile.am:
- use GST_ENABLE_NEW
-
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/resample/private.h:
- don't use optimizations that are #if 0'ed
-
-2004-05-24 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-05-24 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-05-24 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/Makefile.am:
- add rules to build shout2send (was removed by accident
- when this module was no more marked experimental/broken)
-
-2004-05-24 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/shout2/gstshout2.c:
- * ext/shout2/gstshout2.h:
- adding a "connection problem" signal to shout2send
- (fixes #142954)
-
-2004-05-21 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/kio/kioreceiver.cpp:
- * ext/kio/kioreceiver.h:
- fix sign comparison issues
-
-2004-05-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * gst/cdxaparse/gstcdxaparse.c:
- * gst/cdxaparse/gstcdxaparse.h:
- some renaming
- add some checks/sanity
- prepare for seek addition
-
- * sys/sunaudio/gstsunaudio.c:
- remove exported dupe init function
-
-2004-05-21 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2004-05-21 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * m4/as-arts.m4:
- sync with upstream version to fix test on FC2
- readd with -ko to preserve Id header
-
-2004-05-20 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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
-
- * sys/sunaudio/gstsunaudio.c: change category to Sink/Audio
-
- * configure.ac: enable speex plugin for speex 1.1.5+
- * ext/speex/gstspeexenc.c: fix cast warning
-
- * ext/esd/README: fix typo
-
-2004-05-20 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-05-20 David Schleef <ds@schleef.org>
-
- * configure.ac:
- * sys/Makefile.am:
-
-2004-05-20 David Schleef <ds@schleef.org>
-
- * 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:
-
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-05-19 Wim Taymans <wim@fluendo.com>
-
- * gst/law/mulaw-conversion.c: (mulaw_encode):
- Fix overflow bug in ulaw encoding.
-
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- don't unref the event twice
-
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * configure.ac:
- remove -Wno-sign-compare
-
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * configure.ac:
- remove -DG_DISABLE_DEPRECATED. It's not usable without workarounds
- if you want to work against glib 2.2 and 2.4
-
-2004-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/debug/tests.c: (md5_get_value):
- fix segfault on gst-inspect
-
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-05-19 Johan Dahlin <johan@gnome.org>
-
- * ext/dv/gstdvdec.c (gst_dvdec_quality_get_type): Add proper
- ending of the array. Fixes gst-inspect segfault on ppc.
-
-2004-05-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/dirac/gstdiracdec.cc : change category to Codec/Decoder/Video
-
- * m4/a52.m4 : don't fix a test that should fail with current a52dec lib
-
-2004-05-18 David Schleef <ds@schleef.org>
-
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert): Fixes for
- warnings (bugs, actually) noticed by gcc but not forte.
-
-2004-05-18 David Schleef <ds@schleef.org>
-
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: New sunaudiosink
-
-2004-05-18 David Schleef <ds@schleef.org>
-
- * 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)
-
-2004-05-18 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-05-18 David Schleef <ds@schleef.org>
-
- * 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
-
-2004-05-18 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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
-
-2004-05-18 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- remove leftover debugging g_print
-
-2004-05-17 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * m4/a52.m4 : fix compilation with -Wall -Werror
- * m4/libfame.m4 : idem
- * m4/libmikmod.m4 : idem
-
-2004-05-17 Benjamin Otte <otte@gnome.org>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- signal the new tags before giving up the reference
-
-2004-05-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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)
-
-2004-05-17 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * configure.ac : fix compilation of v4l2src with "-Wall -Werror"
- fixes #142664
-
-2004-05-17 Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-05-17 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-05-16 Benjamin Otte <otte@gnome.org>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Hi, I'm a memleak
-
-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
-
-2004-05-16 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/audio/audioclock.c:
- Fix wrong return type (#142205).
-
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstmad.c: (gst_mad_class_init), (gst_mad_init):
- Ignore CRCs by default (fixes #142566).
-
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-05-16 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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
-
-2004-05-16 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_caps), (gst_alsa_get_caps):
- simplify caps
-
-2004-05-16 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-05-16 Benjamin Otte <otte@gnome.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
-
-2004-05-16 Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-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
-
-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.
-
-
-2004-05-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/libvisual/visual.c: (gst_visual_chain):
- add missing visual_audio_analyze
-
-2004-05-14 David Schleef <ds@schleef.org>
-
- * 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().
-
-2004-05-14 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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
-
-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"
-
- * 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)
-
- * gst/cdxaparse/gstcdxaparse.c :
- * gst/cdxaparse/gstcdxaparse.h :
- - partial rewrite using RiffRead (ripped iain's wavparse code)
-
- * gst/rtp/gstrtpL16enc.c : typo
- * gst/rtp/gstrtpgsmenc.c : typo
-
-2004-05-13 Benjamin Otte <otte@gnome.org>
-
- * 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>
-
- * 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
-
-2004-05-11 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Use codec_data property instead of flag1 and flag2 for wma
-
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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
-
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-05-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- Fix some odd cases and fix BE metadata parsing of unicode16 text.
-
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2004-05-10 David Schleef <ds@schleef.org>
-
- * configure.ac: Add prototype Dirac support.
- * ext/Makefile.am:
- * ext/dirac/Makefile.am:
- * ext/dirac/gstdirac.cc:
- * ext/dirac/gstdiracdec.cc:
-
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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)
-
-2004-05-10 Wim Taymans <wim@fluendo.com>
-
- * 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>
-
- * 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.
-
-2004-05-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-05-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-05-06 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_set_property), (theora_enc_get_property):
- Also encode the first frame, cleanup some code.
-
-2004-05-06 Wim Taymans <wim@fluendo.com>
-
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Forward events first before deciding that negotiation was
- not performed.
-
-2004-05-06 Wim Taymans <wim@fluendo.com>
-
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- First process the events before deciding that negotiation
- was not performed.
-
-2004-05-06 Wim Taymans <wim@fluendo.com>
-
- * 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>
-
- * 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)
-
-2004-05-04 Wim Taymans <wim@fluendo.com>
-
- * ext/mpeg2enc/gstmpeg2enc.cc: (gst_mpeg2enc_dispose): call the
- parent dispose function to avoid segfault on destroy.
-
-2004-05-04 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-05-04 Benjamin Otte <otte@gnome.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
-
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am:
- * po/af.po:
- * po/az.po:
- * po/en_GB.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- Updated translations
-
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- refactor/comment code
-
-2004-05-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-05-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_metadata):
- * gst/matroska/matroska-ids.h:
- Basic tag reading support.
-
-2004-04-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-04-30 David Schleef <ds@schleef.org>
-
- * 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:
-
-2004-04-30 Colin Walters <walters@verbum.org>
-
- * ext/alsa/gstalsasink.c (gst_alsa_sink_mmap): Plug a memleak.
-
-2004-04-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_getcaps):
- remove broken nego fix
-
-2004-04-30 Benjamin Otte <otte@gnome.org>
-
- * 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 :)
-
-2004-04-29 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-04-29 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * 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.
-
-2004-04-29 David Schleef <ds@schleef.org>
-
- * sys/oss/gstosselement.c: (gst_osselement_rate_probe_check):
- Add a check for a driver bug on FreeBSD. (bug #140565)
-
-2004-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-04-28 David Schleef <ds@schleef.org>
-
- * gst/speed/gstspeed.c: (speed_parse_caps): Fix caps parsing.
- (bug #140064)
-
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
- Don't probe for playback device if we're a source element. Fixes
- #139658.
-
-2004-04-29 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_chain):
- rewrite buffer offset
-
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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>
-
- * 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
-
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
- 2004-04-26 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (plugin_init):
- set id3mux rank to NONE so it doesn't confuse spider
- require audio/mpeg,mpegversion=1 in id3mux
-
-2004-04-26 Benjamin Otte <otte@gnome.org>
-
- * configure.ac:
- detect faad correctly as non-working if it's indeed non-working
-
-2004-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-04-25 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- fix memleak
-
-2004-04-23 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- audio/x-raw-int with height rules! not. Now it's depth.
-
-2004-04-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/shout/gstshout.c: (gst_icecastsend_change_state):
- fix shoutcast not working (fixes #140844)
-
-2004-04-22 Benjamin Otte <otte@gnome.org>
-
- * 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/
-
-2004-04-22 Benjamin Otte <otte@gnome.org>
-
- * 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.
-
-2004-04-20 David Schleef <ds@schleef.org>
-
- * ext/sdl/Makefile.am: Link against libgstinterfaces, not
- libgstxoverlay. jmmv@menta.net (Julio M. Merino Vidal)
- (bug #140384)
-
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
-
- reviewed by David Schleef
-
- * 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).
-
-2004-04-20 David Schleef <ds@schleef.org>
-
- * gst/realmedia/rmdemux.c: This was supposed to part of the
- last checkin. Same idea.
-
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
-
- reviewed by David Schleef
-
- * configure.ac: bump required gstreamer version to 0.8.1.1
- because of following changes [--ds]
-
- * 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.
-
- * 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
-
- * gst/tags/gstvorbistag.c: (ADVANCE): Likewise
-
- * 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.
-
- * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk):
- Likewise.
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead)
- (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise.
-
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Likewise.
-
- * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA)
- (gst_mpeg2subt_chain_subtitle): Likewise.
-
- * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq)
- (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain):
- Likewise.
-
- * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info):
- Likewise.
-
- * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes):
- Likewise.
-
- * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint):
- Likewise.
-
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- update required version of GStreamer because of GST_TIME_FORMAT
-
-2004-04-20 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c: (plugin_init):
- * sys/xvimage/xvimagesink.c: (plugin_init):
- add debugging categories
-
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-04-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS:
- * po/uk.po:
- Added Ukrainian translation (Maxim V. Dziumanenko)
-
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-18 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-04-17 Iain <iain@prettypeople.org>
-
- * gst/wavparse/gstwavparse.c (gst_wavparse_loop): Allow all events,
- not just EOS.
-
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * configure.ac:
- require mpeg2dec >= 0.4.0
-
-2004-04-17 Benjamin Otte <otte@gnome.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.
-
-2004-04-16 David Schleef <ds@schleef.org>
-
- * sys/ximage/ximagesink.h: Compile fix for FreeBSD. (bug #140268)
- * sys/xvimage/xvimagesink.h: same
-
-2004-04-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Fix GST_ELEMENT_ERROR with (NULL)
-
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add div[3456] as fourccs for DivX 3 (fixes #140137).
-
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: bump nano to 1
-
-=== release 0.8.1 ===
-
-2004-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: releasing 0.8.1, "Comforting Sounds"
-
-2004-04-14 Ronald Bultje <rbultje@ronald.bitfreak.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
-
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_plugin_init):
- lower rank of dvddemux so that it's not used for mpeg playback.
-
-2004-04-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * configure.ac:
- save libs correctly when checking mad
-
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mad/gstid3tag.c: (plugin_init):
- lower rank of id3tag as proposed by Benjamin. Fixes #139926.
-
-2004-04-13 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-04-13 David Schleef <ds@schleef.org>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset): Fix memleak.
- patch from Sebastien Cote (bug #139958)
-
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * examples/gstplay/Makefile.am:
- * examples/gstplay/player.c: (main):
- make the commandline player example use gconf settings
-
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-
-2004-04-12 Christian Schaller <Uraeus@gnome.org>
-
- * sys/oss/gstosselement.c: s/lstat/stat/ from freeBSD, since it can
- be a symlink
-
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_data):
- Handle JUNK chunks inside data section. Prevents warnings.
-
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Fix another codecname mismatch.
-
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Bail out if no filename was given.
-
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-04-09 Benjamin Otte <otte@gnome.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
-
-2004-04-08 David Schleef <ds@schleef.org>
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_register): Change rank to PRIMARY.
-
-2004-04-08 David Schleef <ds@schleef.org>
-
- * gst/colorspace/gstcolorspace.c: Don't advertise a conversion
- we don't support (bug #139532)
-
-2004-04-07 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-04-06 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-04-07 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/gconf/test-gconf.c: (main):
- add missing gst_init
-
-2004-04-07 Benjamin Otte <otte@gnome.org>
-
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
- free the mutexes, too
-
-2004-04-07 Benjamin Otte <otte@gnome.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
-
-2004-04-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstmad.c: (normal_seek):
- fix GST_FORMAT_TIME usage
-
-2004-04-05 David Schleef <ds@schleef.org>
-
- * ext/kio/kiosrc.cpp: Undefine KDE_DEPRECATED so we can use
- a deprecated function (hack!)
-
-2004-04-05 Benjamin Otte <otte@gnome.org>
-
- * ext/esd/esdmon.c: (gst_esdmon_get):
- fix nonterminated vararg and memleak
-
-2004-04-05 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-04-05 Stefan Kost <kost@imn.htwk-leipzig.de>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/ladspa/gstladspa.c: (gst_ladspa_class_init):
- check for broken LADSPA parameters (fixes #138635)
-
-2004-04-05 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps):
- advertise buffer-frames correctly on sinkpads
-
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mad/gstmad.c: add debugging category, comment + cleanups
-
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.net>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * configure.ac:
- fix == in test(1) operator
-
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.net>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * configure.ac:
- fix --export-symblos-regex to a working regex.
-
-2004-04-04 Benjamin Otte <otte@gnome.org>
-
- * sys/oss/.cvsignore:
- add for oss_probe
-
-2004-04-03 Tim-Phillip Müller <t.i.m@zen.co.uk>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-04-02 David Schleef <ds@schleef.org>
-
- * ext/gdk_pixbuf/Makefile.am: Make sure gstgdkanimation.h is
- disted (bug #138914)
-
-2004-04-01 Benjamin Otte <otte@gnome.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.
-
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/riff/riff-media.c:
- fail on error, don't try to set stuff on NULL caps
-
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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/az.po:
-
-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
-
-2004-03-24 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-03-24 Benjamin Otte <otte@gnome.org>
-
- * gst/level/gstlevel.c: (gst_level_init):
- add proxying getcaps function, so level doesn't advertise impossible
- caps
-
-2004-03-24 David Schleef <ds@schleef.org>
-
- * 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)
-
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Fix buffer overflow read error.
-
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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...
-
-2004-03-23 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/typefind/gsttypefindfunctions.c: (ape_type_find),
- (plugin_init):
- Add a monkeysaudio typefind function
-
-2004-03-23 Johan Dahlin <johan@gnome.org>
-
- * 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.
-
-2004-03-22 David Schleef <ds@schleef.org>
-
- * sys/oss/Makefile.am:
- * sys/oss/oss_probe.c: (main), (probe_check), (add_range),
- (check_rate), (add_rate): Rate probing test app.
-
-2004-03-21 Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-03-20 David I. Lehn <dlehn@users.sourceforge.net>
-
- * configure.ac: GST_PACKAGE default: s/GStreamer/GStreamer Plugins/
-
-2004-03-20 Tim-Phillip Müller <t.i.m@zen.co.uk>
-
- reviewed by: Benjamin Otte <otte@gnome.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- terminate gst_event_new_discontinuous correctly (fixes parts of
- #137711)
-
-2004-03-19 David Schleef <ds@schleef.org>
-
- * gst-libs/gst/Makefile.am: Enable xoverlay unconditionally,
- since it doesn't depend on X, and it's part of our ABI.
-
-2004-03-19 Iain <iain@prettypeople.org>
-
- * gst/interleave/deinterleave.c (deinterleave_sink_link): Use the
- is_int in the structure, not the local variable.
-
-2004-03-19 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS:
- * po/af.po:
- adding Afrikaans (Petri Jooste)
-
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- throw error instead of g_critical (#137588)
-
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am:
- * configure.ac:
- dist common and m4 correctly
- * po/sv.po:
-
-2004-03-17 David Schleef <ds@schleef.org>
-
- * pkgconfig/gstreamer-media-info.pc.in: Add Version.
- (bug #137348)
-
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS:
- * po/sv.po:
- adding Swedish translation (Christian Rose)
-
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am: use release.mak
-
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * common/ChangeLog:
- * common/gst-autogen.sh:
- add some explanation about the version detection
- * configure.ac:
- fix X check
-
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: bump nano to 1
-
-=== release 0.8.0 ===
-
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: release 0.8.0, "Pharmaceutical Itch"
-
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- update libtool version
- * gst-libs/gst/media-info/Makefile.am:
- actually use libtool version
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: fix speex detection to work with 1.0 but not 1.1
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2004-03-15 Julien MOUTTE <julien@moutte.net>
-
- * 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:
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * *.c, *.cc: don't mix tabs and spaces
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-03-15 Johan Dahlin <johan@gnome.org>
-
- * *.h: Revert indenting
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot 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.
-
-2004-03-15 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * *.c, *.h: run gst-indent
-
-2004-03-14 Benjamin Otte <otte@gnome.org>
-
- * gst/modplug/gstmodplug.cc:
- * gst/modplug/gstmodplug.h:
- set correct timestamps on outgoing buffers
-
-2004-03-14 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-03-14 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-03-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-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
-
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po/LINGUAS:
- * po/sr.po:
- adding serbian as a language
-
-2004-03-13 Benjamin Otte <otte@gnome.org>
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- return taglist correctly from _get function, don't gst_pad_push it.
- (fixes #137042)
-
-2004-03-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
-
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Don't block during probing...
-
-2004-03-12 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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
-
- * configure.ac:
- use pkg-config for some libraries, falling back to the old .m4 way
- (fixes #131270)
- * m4/libdv.m4:
- removed
-
-2004-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-03-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-03-09 Colin Walters <walters@verbum.org>
-
- * gst/audioconvert/gstaudioconvert.c: Fix typo in width 8
- conversion.
-
-2004-03-09 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init):
- the signals take 2 arguments
-
-2004-03-09 David Schleef <ds@schleef.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):
-
-2004-03-09 Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-03-09 David Schleef <ds@schleef.org>
-
- * 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)
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: bump nano to 1
-
-=== release 0.7.6 ===
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.in: releasing 0.7.6, "There"
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-play.pc.in:
- synchronize the two
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * 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
-
-2004-03-08 Christian Schaller <Uraeus@gnome.org>
-
- * 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>
-
- * sys/oss/gstosssink.c: (gst_osssink_get_delay),
- (gst_osssink_get_time):
- add a warning, IMO this won't get triggered anymore, remove later
-
-2004-03-07 David Schleef <ds@schleef.org>
-
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Added Cinepak
- format (bug #136470)
-
-2004-03-07 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-03-07 Christian Schaller <Uraeus@gnome.org>
-
- * ext/nas/nassink.c:
- A bunch of NAS fixes from Arwed von Merkatz
-
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * 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...
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot 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.
-
-2004-03-06 Christophe Fergeau <teuf@users.sourceforge.net>
-
- * ext/mad/gstmad.c: Fix detection of Xing headers
- * gst/tags/gstid3tag.c: Changes to support TLEN tags
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query):
- debug updates
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * 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
-
-2004-03-05 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
- convert channels correctly. convert correctly to unsigned.
-
-2004-03-05 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state): Check if
- we have a window before clearing it.
-
-2004-03-05 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state): Check if we
- have a window before clearing it.
-
-2004-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-03-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-03-05 Benjamin Otte <otte@gnome.org>
-
- * 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>
-
- * configure.ac:
- fix --with-plugins, don't think it ever worked before
- * gst-plugins.spec.in:
- even more updates
-
-2004-03-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/sdl/sdlvideosink.h:
- * sys/ximage/ximagesink.h:
- * sys/xvideo/xvideosink.h:
- * sys/xvimage/xvimagesink.h:
- Fix for move of gstvideosink.h -> videosink.h.
-
-2004-03-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/xwindowlistener/Makefile.am:
- this is a plugin library, not a library
-
-2004-03-01 David Schleef <ds@schleef.org>
-
- * AUTHORS: Added some names. Add yourself if you're still
- missing.
-
-2004-03-01 David Schleef <ds@schleef.org>
-
- * MAINTAINERS: Add
-
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-plugins.spec.in: clean up spec file
-
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tags/Makefile.am: fix plugin filename
-
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-02-29 Christophe Fergeau <teuf@gnome.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
-
-2004-02-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-02-27 David Schleef <ds@schleef.org>
-
- patches from jmmv@menta.net (Julio M. Merino Vidal)
-
- * 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.
-
-2004-02-27 Benjamin Otte <otte@gnome.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.
-
-2004-02-26 David Schleef <ds@schleef.org>
-
- * 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.
-
-=== release 0.7.5 ===
-
-2004-02-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac: release 0.7.5, "Under The Sea"
-
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-02-25 Andy Wingo <wingo@pobox.com>
-
- * 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/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
- Handle the case where the intersected caps is empty.
-
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/law/mulaw-decode.c: (mulawdec_link):
- * gst/law/mulaw.c: (plugin_init):
- fix mulawdec so it actually works again
-
-2004-02-24 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- reviewed by: David Schleef <ds@schleef.org>
-
- * 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)
-
-2004-02-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- only signal tags for bitrate if they're > 0 (#134894)
-
-2004-02-24 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-24 Johan Dahlin <johan@gnome.org>
-
- * gst-libs/gst/gconf/gconf.c: Add \n to g_print error messages
-
-2004-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * autogen.sh:
- replace test -e with test -x for mkinstalldirs to be more portable.
- (fixes #134816)
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- reported by: Stefan Kost <kost@imn.htwk-leipzig.de>
-
- * gst/audioconvert/gstaudioconvert.c: (plugin_init):
- set rank to PRIMARY
- * gst/volume/gstvolume.c: (plugin_init):
- set rank to NONE
- fixes #134960
-
-2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- * configure.ac:
- export [_]*{gst,Gst,GST}.* symbols from plugins
-
-2004-02-22 Christophe Fergeau <teuf@gnome.org>
-
- reviewed by: Benjamin Otte <otte@gnome.org>
-
- * 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)
-
-2004-02-22 Ramon Garcia <ramon_garcia_f@yahoo.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.
-
-2004-02-20 Andy Wingo <wingo@pobox.com>
-
- * gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
- interleave respectively.
-
- * 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.
-
- * configure.ac: Remove intfloat and oneton, add interleave.
-
- * ext/sndfile/gstsf.c: Handle events better.
-
- * gst/audioconvert/gstaudioconvert.c: Change to support int2float
- and float2int operation. int2float has scheduling problems as
- noted in in2float_chain.
-
-2004-02-20 Benjamin Otte <otte@gnome.org>
-
- * 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.
-
-2004-02-19 Benjamin Otte <otte@gnome.org>
-
- * 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
-
-2004-02-18 David Schleef <ds@schleef.org>
-
- * configure.ac: Move massink to gst-rotten
- * ext/Makefile.am:
- * ext/mas/Makefile.am:
- * ext/mas/massink.c:
- * ext/mas/massink.h:
-
-2004-02-18 David Schleef <ds@schleef.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.
-
-2004-02-18 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-18 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-02-14 Julien MOUTTE <julien@moutte.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)
-
-2004-02-13 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-13 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-13 David Schleef <ds@schleef.org>
-
- * gst/intfloat/gstint2float.c: (conv_f32_s16),
- (gst_int2float_chain_gint16): Change stdint usage to glib types.
-
-2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * configure.ac: use GST_ARCH to detect architecture
-
-2004-02-12 Julien MOUTTE <julien@moutte.net>
-
- * gst/vbidec/vbiscreen.c: Fixing thomasvs fixes. Missing header.
-
-2004-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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)
-
-2004-02-11 David Schleef <ds@schleef.org>
-
- 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):
-
-2004-02-11 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-11 David Schleef <ds@schleef.org>
-
- * ext/alsa/Makefile.am: Fix linking against libgstinterfaces.
- (bug #133886) Noticed by bugs@leroutier.net (Stephane LOEUILLET)
-
-2004-02-11 David Schleef <ds@schleef.org>
-
- * 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
-
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- require gettext 0.11.5 so ulonglong.m4 gets checked out and copied
- by autopoint (fixes #132996)
-
-2004-02-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-02-11 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-02-10 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-02-08 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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)
-
-2004-02-07 David Schleef <ds@schleef.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_handle_event): Don't
- push events to pads that haven't been created (#133508)
-
-2004-02-07 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2004-02-06 David Schleef <ds@schleef.org>
-
- 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:
-
-2004-02-05 David Schleef <ds@schleef.org>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_loop): Revert
- last change, because it Just Doesn't Compile.
-
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- skip undecodable id3v2 tag instead of keeping it
-
-2004-02-05 David Schleef <ds@schleef.org>
-
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Unref leaked buffer. (Noticed by Ronald)
-
-2004-02-05 David I. Lehn <dlehn@users.sourceforge.net>
-
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- Sync requires with other checks. >= vs =.
-
-2004-02-06 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-=== release 0.7.4 ===
-
-2004-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * NEWS: GStreamer Plugins 0.7.4 "For Great Justice" released
- * configure.ac: changed for release
-
-2004-02-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-02-04 David Schleef <ds@schleef.org>
-
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt): Fix a caps
- memleak.
-
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- use correct GST_TAG_ENCODER tag
-
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-02-05 Charles Schmidt <cschmidt2@emich.edu>
-
- reviewed by Benjamin Otte
-
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- extract track count (fixes #133410)
-
-2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_caps_nego):
- that should be !=, not == (fixes #132519)
-
-2004-02-04 David Schleef <ds@schleef.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):
-
-2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * examples/switch/.cvsignore:
- Ignore generated file.
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.c: (gst_play_new):
- check for GError
-
-2004-02-01 Julien MOUTTE <julien@moutte.net>
-
- * 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
-
-2004-02-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * 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
-
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- Fix audio.
-
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegaudioparse/gstmpegaudioparse.c:
- (mp3_type_frame_length_from_header):
- Fix header parsing - stolen from ffmpeg (thank you! :) ).
-
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * just about every source file:
- gst_element_error -> GST_ELEMENT_ERROR
-
-2004-01-29 Julien MOUTTE <julien@moutte.net>
-
- * 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!
-
-2004-01-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpeg1videoparse/gstmp1videoparse.c:
- (gst_mp1videoparse_real_chain):
- Committed wrong version last week... Grr... Didn't notice until now.
-
-2004-01-29 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-01-29 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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...
-
-2004-01-28 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-01-28 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_query_func):
- use gst_element_get_time to get correct time
-
-2004-01-28 Julien MOUTTE <julien@moutte.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.
-
-2004-01-27 David Schleef <ds@schleef.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.
-
-2004-01-27 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_tick_callback),
- (gst_play_seek_to_time): Fixing the way to get current position.
-
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/oss/gstosssink.c: (gst_osssink_sink_query):
- use gst_element_get_time to get correct time
-
-2004-01-27 Julien MOUTTE <julien@moutte.net>
-
- * 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 !
-
-2004-01-27 David Schleef <ds@schleef.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Use gst_pad_try_set_caps_nonfixed().
-
-2004-01-27 David Schleef <ds@schleef.org>
-
- * 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
-
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
- stop processing after EOS
-
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-01-26 Colin Walters <walters@verbum.org>
-
- * 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.
-
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
-
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_extradata):
- Fix SVQ3 decoding on PPC
-
-2004-01-26 Julien MOUTTE <julien@moutte.net>
-
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Dunno how
- that one managed to stay there... Fixed.
-
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
-
- * 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
-
-2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/Makefile.am:
- Fix nas DIST_SUBDIRS
- Uraeus:
- Fix bug where make distcheck doesn't get run on adding stuff to
- the build.
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c: (gst_divxdec_init), (gst_divxdec_setup):
- * ext/divx/gstdivxdec.h:
- Fix divx3 ("msmpeg4") playback using divxdec.
-
-2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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/gnomevfs/gstgnomevfssrc.c:
- * sys/v4l2/v4l2_calls.h:
- fix includes for distcheck
-
-2004-01-21 Christian Schaller <uraeus@gnome.org>
-
- * ext/nas/
- Add libnas (network audio system) plugin, patch from Arwed von Merkatz
- based on earlier patch from Laurent Vivier
-
-2004-01-20 Jeremy Simon <jesimon@libertysurf.fr>
-
- * 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)
-
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_update_streaminfo),
- (gst_vorbisfile_new_link):
- signal streaminfo through tags
-
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- g++ doesn't like NULL in our i18n/error macros, should be
- either (NULL) or ("").
-
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- fix thomas' error errors.
-
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mpeg2enc/gstmpeg2enc.cc:
- fix error errors.
-
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- fix for new error reporting
-
-2004-01-20 David Schleef <ds@schleef.org>
-
- * 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
-
-2004-01-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * configure.ac:
- Up version requirement to 2.0.3 (not yet released) to avoid symbol
- clashes with ffmpeg.
-
-2004-01-20 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/v4l_calls.h:
- * sys/v4l2/v4l2_calls.h:
- element_error fixes
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/gst-i18n-plugin.h:
- add locale.h
- remove config.h inclusion
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2004-01-19 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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>
-
- * configure.ac:
- * ext/Makefile.am:
- * ext/musicbrainz/gsttrm.c:
- * ext/musicbrainz/gsttrm.h:
- * ext/musicbrainz/Makefile.am:
- Add a trm plugin
-
-2004-01-18 Julien MOUTTE <julien@moutte.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.
-
-2004-01-16 David Schleef <ds@schleef.org>
-
- * 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.
-
-2004-01-17 Christian Schaller <Uraeus@gnome.org>
-
- * tools/gst-launch-ext.in - update for new plugins
-
-2004-01-16 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect): Fix use of
- already-freed caps.
-
-2994-01-16 Christian Schaller <Uraeus@gnome.org>
-
- * Update spec for new colorspace plugin and libcaca plugin
- * Fix compilation of libcaca plugin (clock -> id)
-
-2004-01-16 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/Makefile.am:
- restructure so having local patches works easier.
-
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mpeg2enc/Makefile.am:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- Bugfix with respect to EOS handling.
-
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.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.
-
-2004-01-15 David Schleef <ds@schleef.org>
-
- * 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>
-
- * 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>
-
- * 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>
-
- * 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>
-
- * 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>
-
- * 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 :)
-
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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).
-
-2004-01-07 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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
-
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-06 Iain <iain@prettypeople.org>
-
- * gst/intfloat/gstint2float.c (gst_int2float_link): Set the
- buffer-frames caps too.
-
- * 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...
-
-2004-01-05 Ronald Bultje <rbultje@ronald.bitfreak.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).
-
-2004-01-04 David Schleef <ds@schleef.org>
-
- 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):
-
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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
-
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/wavenc/gstwavenc.c: (set_property), (gst_wavenc_init):
- Fix indenting, fix pad creation.
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_loop):
- Finish flac decoder on EOS. See #116178.
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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...
-
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mpeg2dec/gstmpeg2dec.c:
- Add support for mpeg2dec-0.4.0 (released two weeks ago). See
- #130416.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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>
-
- * 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>
-
- * 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
-
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Free XVAdapterInfo correctly.
-
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2003-12-27 Amaury Jacquot <sxpert@esitcom.org>
-
- * ext/ivorbis/vorbis.c:
- * ext/ivorbis/vorbisenc.h:
- * ext/ivorbis/vorbisfile.c:
- Modify so that it uses the new caps things
-
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * testsuite/spider/spider1.c: (main):
- * testsuite/spider/spider2.c: (main):
- * testsuite/spider/spider3.c: (main):
- Make tests compile again. They probably don't work.
-
-2003-12-24 Colin Walters <walters@verbum.org>
-
- * sys/oss/gstosssink.c (gst_osssink_sink_fixate): Return NULL if
- we can't fixate the caps anymore.
-
-2003-12-23 David Schleef <ds@schleef.org>
-
- * gst/volume/gstvolume.c: (volume_init): Proxy getcaps.
- * sys/oss/gstosssink.c: (gst_osssink_init),
- (gst_osssink_sink_fixate): Add fixate function.
-
-2003-12-24 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * 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.
-
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- Fix for 24bpp display.
-
-2003-12-23 Colin Walters <walters@verbum.org>
-
- * ext/gnomevfs/gstgnomevfssink.c: Add ARG_HANDLE property that takes
- a GnomeVFSHandle directly.
-
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/qtdemux/qtdemux.c: (plugin_init):
- qtdemux requires bytestream
-
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_link):
- Fix remaining caps handling errors due to CAPS merge.
-
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * 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.
-
-2003-12-21 Julien MOUTTE <julien@moutte.net>
-
- * 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)
-
-2003-12-21 Julien MOUTTE <julien@moutte.net>
-
- * 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.
-
-2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * 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
projects / platform / upstream / gstreamer.git / blobdiff