-2008-12-11 Edward Hervey <edward.hervey@collabora.co.uk>
+=== release 0.10.30 ===
- * 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
+2010-07-15 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- 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 Wim Taymans <wim.taymans@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.
-
-2008-12-11 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- Pushing 10 buffers is enough to run the test.
-
-2008-12-11 Wim Taymans <wim.taymans@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.
-
-2008-12-10 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- 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.
+ * configure.ac:
+ releasing 0.10.30, "Difficult Birth"
-2008-12-09 Julien Moutte <julien@fluendo.com>
+2010-07-15 00:32:09 +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).
+ * po/cs.po:
+ * po/lv.po:
+ po: update translations
-2008-12-09 Stefan Kost <ensonic@users.sf.net>
+2010-07-14 12:59:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-read.c:
- Fix handling of odd chunks in riff metadata.
+ * 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.
-2008-12-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-07-08 14:51:08 +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.
+ * docs/design/Makefile.am:
+ docs: dist more of the gst-plugin-base design docs
-2008-12-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-07-07 00:35:37 +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.
-
-2008-12-08 Stefan Kost <ensonic@users.sf.net>
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.29.4 pre-release
- * gst/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>
+2010-07-07 00:24:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Olivier Crete <tester at tester ca>
+ * 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-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Implement gst_rtcp_packet_remove(). Fixes #563174.
+2010-07-06 09:47:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add unit test for some RTCP functions.
+ * 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>
-2008-12-04 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Use correct Y offset for YVYU -> RGB conversions
+ Fixes bug #623530.
- * configure.ac:
- Apparently AC_CONFIG_MACRO_DIR breaks when using more
- than one macro directory, reverting last change.
+2010-07-04 17:26:03 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-12-04 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.
- * configure.ac:
- Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
- our M4 macros.
+2010-07-04 14:55:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-12-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * docs/plugins/inspect/plugin-ogg.xml:
+ docs: update ogg introspection info after riff fourcc addition
- * 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
+2010-07-02 20:09:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- Patch by: Cygwin Ports maintainer
- <yselkowitz at users dot sourceforge dot net>
+2010-07-02 14:56:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
-2008-11-28 Christian Schaller <christian.schaller@collabora.co.uk>
+2010-07-02 13:59:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/speexresample/Makefile.am: fix build by removing -lgstaudio
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix invalid memory accesses with odd widths/heights during subsampling
+ Fixes bug #623375.
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-07-01 21:21:38 +0200 Sebastian Dröge <sebastian.droege@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/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.
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-30 21:20:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/speexresample/README:
- Update README with the latest diff between the Speex resampler
- and our copy.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add FLV4 fourcc and map it to video/x-vp6-flash
+ Fixes #623176.
-2008-11-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-30 15:13:10 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/speexresample/gstspeexresample.c: (plugin_init):
- Update the debug category from speex_resample to audioresample.
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: declare with G_DEFINE_TYPE for type safety
+ Fixes #623233.
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-24 16:55:57 +0200 Fredrik Söderquist <fs@opera.com>
- * 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.
+ * 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.
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-30 11:00:45 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/inspect/plugin-audioresample.xml:
- Regenerated for library filename change.
+ * configure.ac:
+ configure: keep things sorted alphabetically
+ On special request. Because it's important, apparently.
-2008-11-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-29 18:48:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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-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:
- * 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.
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.29.3 pre-release
-2008-11-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-29 18:46:18 +0100 Tim-Philipp Müller <tim.muller@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!
+ * configure.ac:
+ configure: fix --disable-external
-2008-11-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-28 15:43:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: 이문형 <iwings at gmail dot com>
+ * autogen.sh:
+ * configure.ac:
+ Bump automake requirement to 1.10
+ For maintainability reasons and $(builddir).
+ Fixes #622944.
- * 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.
+2010-06-27 10:43:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@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.
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- Fix typo in the docs.
+2010-06-26 21:28:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videotestsrc/Makefile.am:
+ videotestsrc: Explicitely link with $(LIBM)
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
- If no stream was found before receiving EOS, post an error message.
- Fixes #561924.
+2010-06-26 21:27:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/Makefile.am:
+ videoscale: Explicitely link with $(LIBM)
- * 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.
+2010-06-26 18:19:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ 0.10.29.2 pre-release
- * gst/playback/gstplaybin2.c: (activate_group):
- Fix buffer-duration property.
+2010-06-26 18:19:33 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * 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.
+2010-06-26 18:19:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 Michael Smith <msmith@songbirdnest.com>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translations
- * 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.
+2010-06-26 17:55:12 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-24 Stefan Kost <ensonic@users.sf.net>
+ * 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
- * 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.
+2010-06-26 12:39:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 David Schleef <ds@schleef.org>
+ * docs/design/design-orc-integration.txt:
+ docs: fix a few typos
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- If we're muxing a dirac stream, flush the page after every picture.
+2010-06-26 12:03:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-24 Stefan Kost <ensonic@users.sf.net>
+ * 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-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.
+2010-06-26 11:38:56 +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.
+ * 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.
- * 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.
+2010-06-25 17:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-11-22 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.
- * gst/videorate/gstvideorate.c:
- Add jpeg and png image media types to the caps. Fixes #561436.
+2010-06-24 13:30:59 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-22 Wim Taymans <wim.taymans@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
- * 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.
+2010-06-24 15:31:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-11-21 David Schleef <ds@schleef.org>
+ * 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.
- Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
+2010-06-24 15:13:31 +0100 Tim-Philipp Müller <tim.muller@collabora.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/check/gst/typefindfunctions.c:
+ tests: add test that runs all typefinders over random data
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-06 12:31:35 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
- * gst/playback/gstdecodebin2.c:
- Fix random fat-fingering making this not compile.
+ * 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
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-24 13:42:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * .gitignore:
+ .gitignore: add temporary orc test directory
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-24 13:30:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Fix win32 build. Oops.
+ * tests/check/Makefile.am:
+ tests: add plugin loading whitelist to test environment
+ Only want to load core/-base plugins here.
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-24 15:09:04 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtsp/gstrtspdefs.c:
- Use WSAGetLastError() rather than errno/h_errno on win32.
+ * common:
+ Automatic update of common submodule
+ From 73ff93a to a519571
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
+2010-06-24 08:41:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c:
- Support WMA Lossless properly.
+ * gst-libs/gst/tag/gsttageditingprivate.c:
+ tag: Fix printf format string
+ Use %s for strings, not %d.
-2008-11-18 David Schleef <ds@schleef.org>
+2010-06-24 12:06:05 +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).
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix resampling of ARGB scanlines
+ Previously we would read behind the end of the source lines.
-2008-11-18 Alessandro Decina <alessandro.d@gmail.com>
+2010-06-16 14:08:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/playback/gstdecodebin.c:
- Add a "sink-caps" property to decodebin like it's done for decodebin2.
- Fixes #560380.
+ * 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.
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-16 11:19:37 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c:
- Guard against a NULL dereference I somehow encountered -
- with a FLUSH_STOP arriving either before basetransform _start(),
- or after _stop().
+ * 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.
- * gst/typefind/gsttypefindfunctions.c:
- Make sure we never jump backwards when typefinding corrupt mov files.
+2010-06-23 12:10:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-11-14 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/playback/gstplaysink.c:
+ playsink: clear ts-offset pointer
+ We need to clear the pointer to our ts-offset element when we destroy the video
+ chain elements to make sure nobody derefs it to invalid memory afterwards.
- * gst-libs/gst/interfaces/propertyprobe.c:
- Fix random type causing a docs warning.
+2010-06-23 10:16:07 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-11-14 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaysink.c:
+ playsink: Reset ts_offset field when freeing chain
+ Otherwise we would end up with a bogus ->audiochain->ts_offset field
+ which would cause segfaults/assertions when trying to modify the
+ 'ts-offset' property in update_av_offset().
+ Was easy to trigger when using a list of audio+video files mixed with
+ video-only files in totem.
- * sys/v4l/gstv4l.c:
- Give it a minimal rank for autovideosrc.
+2010-06-18 16:37:14 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-11-13 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * tests/check/elements/adder.c:
+ * tests/check/elements/appsink.c:
+ * tests/check/elements/audiotestsrc.c:
+ * tests/check/elements/gdpdepay.c:
+ * tests/check/elements/gdppay.c:
+ * tests/check/elements/multifdsink.c:
+ * tests/check/elements/videotestsrc.c:
+ * tests/check/elements/vorbisdec.c:
+ tests: use our own macros for the tests main function
- * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
- (plugin_init):
- Improve typefinding of ISO JPEG2000 mime types.
+2010-06-18 14:17:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * 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 12:27:02 +0200 Philippe Normand <pnormand@igalia.com>
- * 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.
+ * 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
-2008-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-06-17 17:07:39 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c:
- Change author string after seeing output of gst-inspector.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add YUY2/YVYU to all RGB formats conversions
-2008-11-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-17 16:57:43 +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/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y42B to YUY2/YVYU/UYVY conversion for odd widths
-2008-11-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-17 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Fix case where we don't have a range for the rates or channels as is the
- case with truespeech.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix YUY2/YVYU/UYVY to Y42B conversion for odd widths
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-17 16:06:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_update_real_volume),
- (gst_volume_set_volume), (gst_volume_get_volume),
- (gst_volume_set_mute), (gst_volume_init), (volume_setup),
- (volume_transform_ip), (volume_update_mute),
- (volume_update_volume), (volume_get_property):
- * gst/volume/gstvolume.h:
- Keep negotiated state in a separate variable.
- Protect the volume and mute properties with the object lock.
- Protect modifying the transform with the transform lock.
+ * common:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ docs: update introspected plugin docs for gstdoc-scangobj and other changes
+ Update common for latest gstdoc-scangobj and inspect xml files for
+ escaping and pad template order changes. Update other gtk-doc files
+ for API additions and object hierarchy changes.
-2008-11-05 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-16 19:15:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Only convert caps to string when debug is enabled.
+ * 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.
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-15 16:49:17 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 9339ccc to 35617c2
-2008-11-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-15 16:53:49 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 5adb1ca to 9339ccc
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-15 16:34:54 +0300 Stefan Kost <ensonic@users.sf.net>
- Based on patch by: Matthias Kretz <kretz at kde dot org>
+ * common:
+ Automatic update of common submodule
+ From 57c89b7 to 5adb1ca
- * 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
+2010-06-15 15:32:34 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-11-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ Automatic update of common submodule
+ From c804988 to 57c89b7
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+2010-06-15 13:09:37 +0200 Edward Hervey <bilboed@bilboed.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.
+ * tests/check/elements/audioresample.c:
+ Revert "audioresample: set pads as negotiable"
+ This reverts commit 5f74f3a82eb54f9a9517f99dffbe45ce4d474870.
-2008-10-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-15 13:09:29 +0200 Edward Hervey <bilboed@bilboed.com>
- Patch by: Nick Haddad <nick at haddads dot net>
+ * tests/check/elements/audioconvert.c:
+ Revert "audioconvert: set pads negotiable"
+ This reverts commit bbd7dee8f604bd0373a82e6e5cc3eec8313806ac.
- * 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.
+2010-06-14 15:19:32 -0700 David Schleef <ds@schleef.org>
-2008-10-30 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix black horizontal line in image
- * gst/audioresample/gstaudioresample.c:
- Return the result of parent_class->event().
+2010-06-14 15:05:16 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-10-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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/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.
+2010-06-14 18:10:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-10-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/audioconvert.c:
+ audioconvert: set pads negotiable
- * win32/common/libgstaudio.def:
- Add new symbols.
+2010-06-14 17:48:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-10-23 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/audioresample.c:
+ audioresample: set pads as negotiable
- * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
- Attempt to make obfuscated code clearer.
+2010-06-14 16:25:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * common:
+ Revert accidental downgrade of common revision.
- * 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.
+2010-06-14 16:07:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/videoscale.c:
+ videoscale: And only expect a single buffer in the unit test
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.h:
- Don't mark as gtk-doc docs as they aren't public.
+2010-06-14 16:02:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-22 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/elements/videoscale.c:
+ videoscale: Only convert one buffer instead of five
+ Should make the unit test a lot faster.
- * 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
+2010-06-14 14:13:32 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-22 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix unitialized variables
+ yay macosx compilers :(
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Remove useless buffer size assignment. It already has this value.
+2010-06-14 14:13:16 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/video/video.c:
+ video: Fix unitialized variable.
+ yay macosx compilers :(
- * 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.
+2010-06-14 13:27:01 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use Quarks for structure name/field checking
- * gst-libs/gst/audio/gstringbuffer.h:
- Update some docs.
+2010-06-14 13:26:02 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * 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.
+2010-06-14 13:24:06 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Transfer structures instead of copying them
+ Avoids many expensive structure copies
- * tests/icles/.cvsignore:
- update ignore file.
+2010-06-14 13:20:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/Makefile.am:
- * tests/icles/test-box.c: (make_pipeline), (main):
- Add another interactive command line experimentation suite for
- dynamically boxing/cropping/saling an input video.
+ * configure.ac:
+ configure: Use GLIB_EXTRA_CFLAGS
-2008-10-17 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-14 13:02:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
- (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add methods to more accuratly control the pulling thread of a
- ringbuffer.
- Add format conversion helper code to the ringbuffer.
- API: GstRingBuffer:gst_ring_buffer_activate()
- API: GstRingBuffer:gst_ring_buffer_is_active()
- API: GstRingBuffer:gst_ring_buffer_convert()
+ * common:
+ Automatic update of common submodule
+ From 7a0fdf5 to c804988
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-14 11:31:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_stop):
- Signal thread startup earlier so that we can immediatly go into pull
- mode when we have to and block on preroll.
+ * common:
+ Automatic update of common submodule
+ From 6da3bab to 7a0fdf5
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-14 11:20:00 +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.
+ * tests/examples/seek/jsseek.c:
+ jsseek: Don't use deprecated GLib API
+ Fixes once again bug #605100.
-2008-10-16 Stefan Kost <ensonic@users.sf.net>
+2010-06-14 11:16:45 +0200 Prahal <prahal at yahoo.com>
- * 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.
+ * 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
-2008-10-16 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-14 11:11:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-14 11:08:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add mappping for the KMVC (Karl Morton's Video) Codec.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ gnomevfssrc: Fix possible NULL pointer dereference
+ It's always an error if gst_buffer_try_new_and_alloc() returns NULL
-2008-10-15 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-14 11:03:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Return FALSE from the seek handler if no seek callback was set
-2008-10-15 Sebastian Dröge <slomo@circular-chaos.org>
+2010-06-14 09:53:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/gio/gstgiostreamsrc.c:
+ giostreamsrc: Fix copy&paste error in the docs
-2008-10-14 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-06-14 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_decode_buffer):
- Fix build on macosx.
+ * 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'".
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-13 22:17:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Based on patch by: Robin Stocker <robin at nibor dot org>
+ * tests/check/elements/videoscale.c:
+ videoscale: Add ffmpegcolorspace after videotestsrc for the unit test
- * 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.
+2010-06-13 20:57:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: ...and add Y16 case for the linear scaling
- * 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.
+2010-06-13 20:38:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps):
- Propagate return value of setcaps.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add Y16 case for 4-tap scaling
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-13 18:27:19 +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.
+ * tests/check/Makefile.am:
+ tests: Fix linking of the tags test
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-13 08:20:13 +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.
+ * gst-libs/gst/video/video.h:
+ video: Re-add (but deprecated) GST_VIDEO_{RED,GREEN,BLUE}_MASK_1[56]
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-12 21:04:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- For looking at the 4th byte we have to get 4 bytes of course
- and not 3.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use correct variables for debug output
-2008-10-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-12 16:51:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
- Improve FLAC-without-headers typefinding by looking at most of the
- frame header and checking if invalid values are used. Should prevent
- quite some false positives compared to the old version which only
- check if the first 14 bits are set.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y16 from/to GRAY8 conversion
-2008-10-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-12 16:31:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- Don't assert on caps==NULL.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Don't crash when doing gray YUV to GRAY conversion
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-12 16:23:23 +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.
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.h:
+ videoscale: Update disted orc files
-2008-10-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-12 16:16:37 +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.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Allow video/webm for progressive downloading
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-12 13:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Based on a patch by: xavierb at gmail dot com
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for more gray formats
- * 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.
+2010-06-01 16:45:34 +0000 Martin Bisson <martin.bisson@gmail.com>
-2008-10-10 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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.
- * ext/vorbis/vorbisenc.c:
- (gst_vorbis_enc_buffer_check_discontinuous):
- Fix discontinuity detection which was broken by last commit.
+2010-06-01 14:42:54 +0000 Martin Bisson <martin.bisson@gmail.com>
-2008-10-09 Tim-Philipp Müller <tim.muller at 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.
- * configure.ac::
- Require core CVS for ghostpad API additions used by decodebin2.
+2010-06-12 13:35:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-08 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Use libgstvideo for caps parsing, etc
- * 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.
+2010-06-12 13:04:43 +0200 Philippe Normand <phil@base-art.net>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix format string compiler warning on OS X
- * 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.
+2010-06-12 13:00:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use GST_VIDEO_CAPS_GRAY{8,16}
- Patch by: Pavel Zeldin <pzeldin at gmail dot com>
+2010-06-12 12:57:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/gstvideoscaleorc.orc:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Implement linear merging of Y16 scanlines with orc
-2008-10-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-12 08:26:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Jan Gerber <j at oil21 dot org>
+ * common:
+ Automatic update of common submodule
+ From 733fca9 to 6da3bab
- * 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.
+2010-06-11 22:16:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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).
- Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+2010-06-11 12:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Implement skew clock slaving. Fixes #552559.
+ * 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.
-2008-10-08 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-10 18:12:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- * gst-libs/gst/audio/testchannels.c:
- Fix include of config.h
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add typefinder for dts audio
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-11 15:23:14 +0200 Edward Hervey <bilboed@bilboed.com>
- Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix unitialized variables
+ I hate thee macosx
- * 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.
+2010-06-11 08:47:27 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/tag/gstexiftag.c:
+ gstexiftag: Fix debug statements
- * 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.
+2010-06-11 08:47:17 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-10-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/tag/gstexiftag.c:
+ exiftag: Fix unitialized variable
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_chain):
- Improve debugging of the rtptime.
+2010-06-10 20:45:42 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-10-05 Jan Schmidt <jan.schmidt@sun.com>
+ * win32/common/libgsttag.def:
+ win32: update def file
- * configure.ac:
- Back to development -> 0.10.21.1
+2010-06-10 20:36:32 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/tag/tag.h:
+ docs: fix gtk-doc warnings
+ Variable names in function prototypes in the headers should match the doc-
+ comment.
- * 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.
+2010-06-10 08:47:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: If the text-sink claims to support ANY caps assume it only support raw plaintext subtitles
+ Fixes bug #621071.
- * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
- (plugin_init):
- Add typefinder for MXF.
+2010-06-10 08:46:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+ * tests/icles/playbin-text.c:
+ icles: Only accept plain subtitles in the playbin-text icles test
- * tests/icles/Makefile.am:
- Only build test-colorkey if GTK+ is available.
+2010-06-09 22:34:24 +0200 Edward Hervey <bilboed@bilboed.com>
-=== release 0.10.21 ===
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for VP6F (On2 VP6 Flash variant)
-2008-10-03 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-09 12:35:40 -0700 David Schleef <ds@schleef.org>
* configure.ac:
- releasing 0.10.21, "We Could Be Wrong"
+ Use the Orc m4 macro
-2008-09-28 Jan Schmidt <jan.schmidt@sun.com>
+2010-06-09 12:40:00 -0700 David Schleef <ds@schleef.org>
- * configure.ac:
- 0.10.20.4 pre-release
+ * common:
+ Automatic update of common submodule
+ From fad145b to 733fca9
-2008-09-25 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-09 12:33:51 -0700 David Schleef <ds@schleef.org>
- Patch by: Vincent Penquerc'h <ogg dot k dot ogg dot k at googlemail dot com>
+ * common:
+ Automatic update of common submodule
+ From 47683c1 to fad145b
- * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
- Set the BOS flag on the BOS packet. Fixes #553244.
+2010-06-09 15:58:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-23 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * gst-libs/gst/rtsp/gstrtspmessage.c:
- (gst_rtsp_message_parse_request),
- (gst_rtsp_message_parse_response):
- Fix the g_return_val_if_fail() statements.
+2010-05-10 14:01:46 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-09-22 Michael Smith <msmith@songbirdnest.com>
+ * 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>
- * 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
+ * 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
-2008-09-15 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-03 23:02:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c:
- * gst/typefind/gsttypefindfunctions.c:
- Recognise Kate subtitle streams (#550582).
+ * 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>
-2008-09-13 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle SEEKING query in push mode too
- * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
- Remove trailing comma from enum list, which causes problems
- with -pendantic (#550729).
+2010-06-09 16:38:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Update the total time from the Skeleton 4 indexes
+ Fixes bug #620939, see bug #607945.
- * configure.ac:
- 0.10.20.2 pre-release
+2010-06-09 16:33:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/LINGUAS:
- * po/id.po:
- * po/pt_BR.po:
+ * ext/ogg/gstoggstream.c:
+ oggstream: Implement latest version of the Skeleton 4.0 spec
+ Fixes bug #620939.
- New translations.
+2010-06-09 16:59:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- * 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.
+2010-06-08 13:34:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-05 Stefan Kost <ensonic@users.sf.net>
+ * .gitignore:
+ .gitignore: add orc-related temp files
- * gst-libs/gst/interfaces/propertyprobe.c:
- Check for NULL pointer, in the hope that this fixes #532864.
+2010-06-08 13:26:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-09-05 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- No really, the next release is 0.10.21 (fix Since: tags in docs).
+2010-06-08 13:11:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-04 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Implement correct parsing of Skeleton 4.0 index packets
- * 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.
+2010-06-08 12:01:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-04 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Add parsing of Skeleton 4.0 indexes
- * 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.
+2010-06-08 11:40:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * 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-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.
+2010-06-08 11:31:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Edward Hervey <edward.hervey@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: Parse Skeleton stream major/minor version
- * 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
+2010-06-08 11:26:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Use binary search for searching in the index
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
- Fixes #550638.
+2010-06-08 11:02:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-03 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/libs/video.c:
+ video: Fix unit test, the Y800 checks were not used before and were not working
- * 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.
+2010-06-08 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/video/video.c:
+ video: Return TRUE in gst_video_format_is_gray() for Y800 and Y16
- * configure.ac:
- Disable subparse when xml is disabled. It woundn't work anyway. Fixes
- test runs.
+2010-06-08 00:33:31 -0700 David Schleef <ds@schleef.org>
-2008-09-02 Tim-Philipp Müller <tim.muller at 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.
- * po/POTFILES.in:
- Add some more files with strings for translation.
+2010-06-08 00:32:36 -0700 David Schleef <ds@schleef.org>
-2008-09-02 Stefan Kost <ensonic@users.sf.net>
+ * REQUIREMENTS:
+ requirements: change liboil to Orc
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c:
- Use new geo location tags from core. Fixes #481169
+2010-06-06 23:50:05 -0700 David Schleef <ds@schleef.org>
-2008-09-01 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: convert from liboil to orc
- * 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.
+2010-06-06 23:48:35 -0700 David Schleef <ds@schleef.org>
-2008-08-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * tests/check/Makefile.am:
+ tests: Add orc tests
- Patch by: Jonathan Matthew <notverysmart gmail com>
+2010-06-06 23:48:15 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-08-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:47:53 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:47:16 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-08-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-06 23:46:41 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-06 23:45:58 -0700 David Schleef <ds@schleef.org>
- Patch by: Mike Ruprecht <cmaiku at gmail dot com>
+ * 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
- * 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.
+2010-06-06 23:45:10 -0700 David Schleef <ds@schleef.org>
-2008-08-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * docs/design/Makefile.am:
+ * docs/design/design-orc-integration.txt:
+ docs: Add notes about Orc integration
- Patch by: Alessandro Dessina <alessandro nnva org>
+2010-06-06 23:34:39 -0700 David Schleef <ds@schleef.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.
+ * configure.ac:
+ configure: convert liboil check to orc
-2008-08-22 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-06-08 07:34:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
- Reset variable so that query and convert fail after going back to
- READY. Fixes #548898.
+ * ext/ogg/gstoggmux.c:
+ oggmux: Start a new page for every CMML buffer
-2008-08-22 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-07 14:38:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- If a buffer arrives with a timestamp before the timestamp+duration
- of the previous buffer clip it instead of dropping it completely.
- Slight improvement for the unfixable bug #548913.
+ * 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
-2008-08-21 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-07 08:31:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Take the current timestamp instead of timestamp+duration for the offset.
- This offset will later be used for calculating the timestamp and
- otherwise vorbisdec will interpolate timestamps wrong if upstream
- only sends timestamps and no granulepos.
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: Map "Y8 " and "GREY" to "Y800" and add it to the template caps
-2008-08-21 Stefan Kost <ensonic@users.sf.net>
+2010-06-07 08:17:13 +0200 Martin Bisson <martin.bisson@gmail.com>
- * tests/examples/seek/seek.c:
- Don't crash when having no visualisations.
+ * 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.
-2008-08-16 David Schleef <ds@schleef.org>
+2010-06-07 08:16:01 +0200 Martin Bisson <martin.bisson@gmail.com>
- * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
- check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
- Fixes #548065.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for Y800 and Y16
+ Fixes bug #620441.
-2008-08-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-06-06 16:46:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- When cleaning up the caps fields also remove "depth" for the same
- reason we remove "width".
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix log function printf format issue
-2008-08-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-06-05 18:14:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
- Add Lead H.264 here as well.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: stop jpeg typefinding once we found a SOF marker
-2008-08-14 Julien Moutte <julien@fluendo.com>
+2010-06-05 18:05:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps): Add Lead H.264 variant.
+ * tests/check/gst/typefindfunctions.c:
+ tests: fix memory leak in unit test
-2008-08-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-19 15:40:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- When not slaved to another clock also subtract the base_time from our
- internal clock time to get the running time.
+ * gst/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.
-2008-08-12 David Schleef <ds@schleef.org>
+2010-06-05 17:22:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
- since it has no basis in libtheora.
+ * 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
-2008-08-12 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 16:58:50 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/interfaces/propertyprobe.h:
- Remove double "interface" from doc-string.
+ * 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.
- * gst-libs/gst/interfaces/xoverlay.h:
- Document interface.
+2010-06-05 12:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff.c:
- Add basic doc blobs.
+ * gst/playback/gstscreenshot.c:
+ playbin2: improve screenshot code
+ Use appsrc and appsink in the screenshot code to make things nicer.
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 11:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/Makefile.am:
- Don't try to build that example anymore.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix documentation string
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 11:05:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/.cvsignore:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- * gst-libs/gst/audio/make_filter:
- Move audiofiltertemplate to gst-template.
+ * gst/playback/gstplaysink.c:
+ playsink: add convert-frame action signal
+ Add a convert-frame action signal.
+ Fixes #620279
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 11:02:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
+2010-06-05 10:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
-
-2008-08-11 Stefan Kost <ensonic@users.sf.net>
-
- * gst-libs/gst/audio/audio.c:
- Fix doc comment syntax.
+ * 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
- * gst-libs/gst/interfaces/propertyprobe.c:
- Add more doc-comments and a FIXME: for the signal.
+2010-06-05 10:31:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-08-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/playback/gstplaysink.c:
+ playsink: add property to get the last frame
+ Add a property to get the last video frame.
+ See #620279
- * 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>
+2010-06-04 19:30:14 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/pbutils/descriptions.c:
- Add audio/x-qdm for qtdemux.
+ * 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.
-2008-08-05 Stefan Kost <ensonic@users.sf.net>
+2010-06-03 13:44:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisdec.c:
- Do not leak old taglist.
+ * 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.
-2008-08-04 Stefan Kost <ensonic@users.sf.net>
+2010-06-02 12:19:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/test-scale.c:
- Include <stdlib.h> for atoi().
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: Allocate and free the clock in NULL->READY and reverse
-2008-08-04 Andy Wingo <wingo@pobox.com>
+2010-06-01 23:49:07 -0700 David Schleef <ds@schleef.org>
- * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
- documentation fix.
+ * common:
+ Automatic update of common submodule
+ From 17f89e5 to 47683c1
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+2010-06-01 22:54:33 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From fd7ca04 to 17f89e5
-2008-08-01 Stefan Kost <ensonic@users.sf.net>
+2010-06-01 13:00:22 +0100 Tim-Philipp Müller <tim.muller@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.
+ * 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.
-2008-08-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-31 19:28:45 +1000 Jonathan Matthew <jonathan@d14n.org>
- * 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'.
+ * ext/gio/gstgiobasesink.c:
+ gio: map GIO NO_SPACE error to NO_SPACE_LEFT
+ Fixes bug #620140.
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-28 08:27:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
- Add some more comments.
+ * configure.ac:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ configure: Remove (now) useless check for cbrt
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-12-02 22:16:22 -0800 David Schleef <ds@schleef.org>
- * 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/interfaces/streamvolume.c:
+ interfaces: Use pow() instead of cbrt() for MSVC
-2008-07-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-26 11:54:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-scale.c: (make_pipeline), (main):
- Add dynamic rescaling tests for the new basetransform.
+ * common:
+ Automatic update of common submodule
+ From 357b0db to fd7ca04
-2008-07-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-05-26 08:51:09 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/audioconvert/Makefile.am:
- Dist recently-added gstfastrandom.h.
+ * 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
-2008-07-30 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-05-24 11:27:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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-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).
-2008-07-30 Stefan Kost <ensonic@users.sf.net>
+2010-05-22 10:05:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-ids.h:
- Adding acid chunk for tempo and loop information.
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate interlaced, chroma-site and color-matrix fields if necessary
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2010-05-22 10:02:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/Makefile.am:
- floor() needs linking to $(LIBM).
+ * configure.ac:
+ * gst/videorate/gstvideorate.c:
+ videorate: Use new string fixation function from core
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2010-05-22 09:48:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssrc.c:
- Aggregate short reads and add some comments and debug logging.
- Fixes #537380
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate color-matrix and chroma-site fields if necessary
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+2010-05-22 09:39:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c:
- Fix property doc markup (its not a signal).
+ * gst/videorate/gstvideorate.c:
+ videorate: Fixate the interlaced field if necessary
+ Fixes bug #619310.
- * sys/xvimage/xvimagesink.c:
- Add since tag for new proeprties (also add sice tags fro the last two
- other additions).
+2010-05-22 08:55:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-29 Stefan Kost <ensonic@users.sf.net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add IVF typefinder
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add autofill/colorkey properties. Fixes #538656.
+2010-05-21 18:16:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-07-28 David Schleef <ds@schleef.org>
+ * gst/videorate/gstvideorate.c:
+ videorate: pass object to logging functions, use GST_DEBUG_FUNCPTR
- * 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.
+2010-05-20 15:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add basic descriptions for new WebM and VP8 types
- * 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.
+2010-05-20 14:21:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix sizes again, this time for real
- * ext/ogg/gstoggmux.c:
- Allow muxing of CELT into Ogg streams.
+2010-05-20 13:58:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix size checks
- * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
- (plugin_init):
- Add simple typefinder for the CELT codec (www.celt-codec.org).
+2010-05-20 10:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-27 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * 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
- Patch by: Jan Gerber <j at oil21 dot org>
+2010-05-20 08:52:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
- Fix calculation of the start time from skeleton streams.
- Fixes bug #530068.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Some more minor adjustments for the VP8 Ogg mapping
-2008-07-24 Stefan Kost <ensonic@users.sf.net>
+2010-05-19 21:35:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c:
- Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Update to the latest VP8 mapping
-2008-07-23 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-10 05:53:22 +0200 Philip Jägenstedt <philipj@opera.com>
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_setup_dither),
- (gst_audio_quantize_free_dither):
- * gst/audioconvert/gstfastrandom.h:
- Implement a linear congruential generator as pseudo random number
- generator for the dither noise. This is about 2 times faster than
- using GLib's mersenne twister. Also this uses only integer math for
- generating integers while GLib internally uses floating point math.
+ * gst/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.
-2008-07-23 Michael Smith <msmith@songbirdnest.com>
+2010-05-14 13:31:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Fix granulepos->key granule calculation for Dirac
-2008-07-23 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-05-14 11:02:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Damien Lespiau <damien.lespiau gmail com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Add support for mapping specific granulepos to key granule mapping
- * 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.
+2010-05-05 13:59:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ ogg: Implement Ogg VP8 mapping
- * 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
+2010-04-27 15:24:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-07-17 Jan Schmidt <Jan.Schmidt@sun.com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for On2 VP8
- * 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):
+2010-05-19 16:17:19 +0200 Alessandro Decina <alessandro.decina@collabora.co.uk>
- Apparently on Solaris and OS/X (at least), it's not legal to ask the
- X server to attach to a shared memory segment after we've deleted it,
- with the result that MIT-SHM is disabled. Instead, remove it only after
- X succeeds in attaching too.
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix a typo introduced by 9d753824.
+ video/x-raw-float => audio/x-raw-float. Fixes #619090.
-2008-07-16 David Schleef <ds@schleef.org>
+2010-05-18 08:45:52 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Add 'ticks', a 1/30 second sine wave pulse every second.
+ * 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.
-2008-07-15 David Schleef <ds@schleef.org>
+2010-05-15 21:15:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/video.c: Revert ABI change.
+ * configure.ac:
+ configure: Use = instead of == in shell scripts for equality checks
-2008-07-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-14 18:23:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Make it impossible to have NULL caps at the point where we set
- framerate and other things. Also don't return immediately for "3ivd"
- video and let framerate, etc be set. Might fix bug #542508.
+ * common:
+ Automatic update of common submodule
+ From 4d67bd6 to 357b0db
-2008-07-14 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+2010-05-14 17:24:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
- Video format can also be conveniently determined from (many)
- non-fixed caps.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Always take the object lock when accessing the caps
+ Fixes bug #618625.
-2008-07-14 Jan Schmidt <thaytan@noraisin.net>
+2010-05-14 17:17:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c:
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreamselector.c:
- First stab at integrating DVD subpicture overlay into
- playbin. Successfully plugs and plays, but the queues need
- shrinking - 3 seconds of video is too much buffering.
+ * 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.
-2008-07-11 Stefan Kost <ensonic@users.sf.net>
+2010-05-14 17:13:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Remove now obsolete note in the docs.
+ * gst/playback/gstplaysink.c:
+ playsink: Fix deadlock caused from an additional lock instead of unlock
+ Also improve debug output for the playsink lock.
-2008-07-11 Stefan Kost <ensonic@users.sf.net>
+2010-05-13 12:16:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ 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.
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-13 12:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/Makefile.am:
- Fix out of tree build by adding all required CFLAGS.
+ * 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
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-13 08:05:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/video/video.c:
+ video: Use simple fraction multiplication functions instead of going through GValues
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a unit test for checking if the negotiation works as expected
-2008-07-07 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2010-05-10 17:09:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try harder to keep the DAR if possible
+ Fixes bug #371108.
- * 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.
+2010-05-10 15:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- decodebin2 doesn't have this issue because all streams of a group
- go through multiqueue.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Log PAR and DAR of input and output caps when setting caps
-2008-07-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-05-10 14:52:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Set input width/height if the output caps don't have any width or height
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Makes libgstsdp compile with mingw32 by defining the right WINVER so
- that getaddrinfo() can be used. Fixes #541358.
+2010-05-10 13:01:44 +0200 Andoni Morales <ylatuya@gmail.com>
-2008-07-01 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Try to keep DAR when scaling
+ Fixes bug #371108.
- * 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.
+2010-05-10 19:09:28 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2008-06-30 Sebastian Dröge <sebastian.droege@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
- * 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.
+2010-05-10 18:53:29 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ basertpaudiopayload: Set duration on buffers
+ Set the duration of the buffers from their size
- * 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.
+2010-05-11 16:12:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-30 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Fixate PAR to 1/1 if possible
- * 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.
+2010-05-11 10:07:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-29 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0
- * gst/adder/gstadder.c:
- Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
+2010-05-10 12:44:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * ChangeLog:
- ChangeLog surgery.
-
- * tests/examples/seek/seek.c:
- Move variable into ifdef too.
-
+2010-05-10 12:28:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-06-27 Stefan Kost <ensonic@users.sf.net>
+ * sys/ximage/ximagesink.c:
+ ximagesink: Post an error message on the bus if no supported pixmap formats can be found
+ Might fix bug #615851.
- * tests/examples/seek/seek.c:
- Include config.h and check if we have X. Remove XInitThread(), don't
- think its needed. Fixes: #540334.
+2010-05-07 19:49:57 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-06-26 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace : whooops
- Patch by: Sam Morris <sam at robots dot org to uk>
+2010-05-07 19:21:13 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: more minor cleanups
- * 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.
+2010-05-07 17:16:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-06-25 Stefan Kost <ensonic@users.sf.net>
+ * 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)
- * tests/examples/seek/Makefile.am:
- * tests/examples/seek/seek.c:
- Remove libgstvideo usage. Use gtk_get_option_group instead of
- gtk_init().
+2010-05-07 12:19:25 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add the 'expose-all-streams' property from decodebin2
+ API: expose-all-streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
- * tests/check/Makefile.am:
- Name the test registry format neutral.
+2010-05-06 18:50:51 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+ * 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>
- * gst/playback/gstqueue2.c:
- Do not double notify. Remove the unsued return value.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: rename are_raw_caps to are_final_caps, correct comment
+ https://bugzilla.gnome.org/show_bug.cgi?id=617868
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2010-05-07 17:16:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/sdp/gstsdpmessage.h:
+ sdpmessage: add new TIAS bandwidth modifier
+ Add TIAS modifier as specified in RFC 3890.
+ Do some whitespace fixes.
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2010-05-07 00:10:22 +0300 Stefan Kost <ensonic@users.sf.net>
- * ChangeLog:
- Forgot to save the ChangeLog :/
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: disambigue comment due to popular demand
+ Write "target depth" instead of "our depth" or previous ambigous "out depth".
-2008-06-24 Stefan Kost <ensonic@users.sf.net>
+2010-05-06 15:40:34 +0200 Mark Nauwelaerts <mark.nauwelaerts@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: disconnect signals in some more cleanup cases
-2008-06-24 Jan Schmidt <jan.schmidt@sun.com>
+2010-05-06 13:10:54 +0200 Wim Taymans <wim.taymans@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.
+ * 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.
-2008-06-22 Stefan Kost <ensonic@users.sf.net>
+2010-05-06 12:21:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c:
- Fix crasher when playing a parse-launch line the 2nd time.
+ * 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.
-2008-06-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2010-05-06 12:06:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/pipelines/oggmux.c:
- Properly ifdef tests to fix compilation.
+ * 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.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-05-06 10:56:21 +0200 Wim Taymans <wim.taymans@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.
+ * 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.
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-05-06 11:34:53 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin2.c:
- Use a different constant for the convert-frame signal id.
- Fixes #537009.
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ docs: be more firendly to gtk-doc limitted parsing capabilities
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-05-06 09:42:02 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin2.c:
- * gst/playback/gstplaysink.c:
- Fix a whole bunch of typos in comments and log statements.
+ * 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
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
+2010-05-06 09:17:33 +0300 Stefan Kost <ensonic@users.sf.net>
- * sys/xvimage/xvimagesink.c:
- Don't set colour balance values on the Xv port if the user hasn't
- changed them (via properties or the interface). Avoids accumulating
- rounding errors for the common case.
- Partial fix for bug #537889.
-
-2008-06-20 Michael Smith <msmith@songbirdnest.com>
-
- * gst/playback/gstdecodebin2.c:
- Ensure decodebin2 emits 'drained' signal once, and only once, when all
- pads are drained.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
- (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
- Report the encoder latency. Fixes #538232.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
- (notify_source), (activate_group):
- Implement the source property, emit notify when it changes in the
- underlying uridecodebin.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/examples/seek/seek.c: (stop_cb):
- Free and clear the seek element list so that we don't use invalid
- references when seeking after recreating a gst-launch line.
-
-2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_render):
- Report latency even if we are not live instead of hiding it.
- Take ts-offset and render-delay of the basesink into account when
- scheduling samples.
- Rework the clipping code so that we can take the various offsets into
- account and still do correct clipping.
-
-2008-06-20 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- Bump verion back to devel -> 0.10.20.1
-
-2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
- Don't increase the size of non-string image buffers by one as this
- might in theory confuse decoders. Still increase it by one for string
- image buffers to append '\0'.
-
-2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
-
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
- Fix a buffer memleak and remove a confusing and wrong debug output.
- Fixes bug #538663.
-
-=== release 0.10.20 ===
-
-2008-06-18 Jan Schmidt <jan.schmidt@sun.com>
-
- * configure.ac:
- releasing 0.10.20, "Here I Go Again"
-
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: use can_intersect to avoid a caps copy
- * configure.ac:
- 0.10.19.3 pre-release
+2010-05-06 09:14:25 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-06-11 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/videorate/gstvideorate.c:
+ videorate: trucate own caps, instead of copying and using the first only
+ We got the caps from an intersect, it is our own, hence we can truncate it.
+ Besides gst-indent has chooses to line-up all caps in one line again :/.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Fix build on win32.
- Patch By: David Schleef <ds@schleef.org>
- Fixes: #536874
+2010-05-06 09:12:32 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-06-11 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstdecodebin.c:
+ decodebin: use can_intersect to avoid a caps copy
- * 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.
+2010-05-06 09:11:17 +0300 Stefan Kost <ensonic@users.sf.net>
- 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.
+ * 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.
- Fixes bug #536849 and #536848.
+2010-05-06 08:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-06-05 Jan Schmidt <jan.schmidt@sun.com>
+2010-05-06 08:16:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- 0.10.19.2 pre-release
+ * gst/audioconvert/audioconvert.c:
+ audioconvert: fix typo in comment
-2008-06-04 Jan Schmidt <jan.schmidt@sun.com>
+2010-05-06 08:15:16 +0300 Stefan Kost <ensonic@users.sf.net>
- * win32/common/libgstrtsp.def:
- * win32/common/libgsttag.def:
- Add new API functions to the dll exports
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: use gst_caps_can_intersect() more
+ In place where we just need to know whether caps intersect, we can use this
+ quicker function.
-2008-06-04 Michael Smith <msmith@songbirdnest.com>
+2010-04-15 13:09:45 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-15 13:08:39 +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).
+ * tests/icles/audio-trickplay.c:
+ example: update status (adder is fixed now)
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-04-15 13:08:01 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/check/Makefile.am:
- Do not try to run the check tests for subparse unless it has been
- built.
+ * tests/icles/playbin-text.c:
+ example: make app static
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-05-05 13:25:02 +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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: printf format fixes
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
+2010-05-04 15:32:59 +0100 Tim-Philipp Müller <tim.muller@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.
-
-2008-06-04 Peter Kjellerstedt <pkj@axis.com>
-
- * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
- Fixed accidental use of IPv4 options for all IPv6 addresses.
-
-2008-06-04 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * gst-libs/gst/interfaces/mixertrack.h:
- Document mixer track flags.
-
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
-
- 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.
+ 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.
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+2010-05-04 17:54:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2008-06-04 Sebastian Dröge <slomo@circular-chaos.org>
+2010-05-04 17:38:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: handle corrupt indexes
+ Make sure we handle and receover from corrupt indexes.
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-05-04 15:47:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Based on patch by: John Millikin <jmillikin gmail com>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix EOS check
- * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
- (gst_vorbis_tag_add_coverart):
- Retrieve COVERART tags from vorbis comments (#512333)
+2010-05-04 13:51:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoild division by 0
- * 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...).
+2010-05-04 13:50:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-06-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: cleanup unused defines
- * 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
+2010-05-04 13:36:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * win32/common/libgstaudio.def:
- Add gst_audio_check_channel_positions() to the exported symbols.
+2010-05-04 13:05:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-06-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use skeleton duration when possible
- * 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.
+2010-05-04 13:02:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/libs/audio.c: (GST_START_TEST):
- Add some simple checks for gst_audio_check_channel_positions().
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggstream: parse duration from 3.3 skeleton
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-02 11:16:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- minrange and maxrange are scaled according to the frequency
- multiplier.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: more index parsing work
-2008-06-02 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-01 13:50:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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).
- * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
- (main):
- Test with odd height/width too.
+2010-05-04 11:19:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-06-02 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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/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.
+2010-05-03 16:52:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-31 Mark Nauwelaerts <mnauw@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-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.
+2010-04-30 18:37:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-31 Mark Nauwelaerts <mnauw@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.
- * 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>
+2010-04-30 18:03:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: refactor for seeking in pushmode
+ refactor the code a little to prepare for seeking in push mode
- * win32/common/libgstaudio.def:
- Add gst_audio_clock_reset to the list of exported symbols.
+2010-05-03 12:46:34 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
- Remove wrong_channels_identification_header unit test as we now
- support 7 (and more channels).
+2010-04-26 22:08:41 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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/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.
+2010-04-30 19:56:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- * 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.
+2010-04-30 19:37:33 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Allow setting channel positions for 1/2 channels when using
- gst_audio_set_structure_channel_position().
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: ensure writable metadata
- * 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.
+2010-04-30 17:41:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- We now allow previously conflicting channel positions to appear
- together (rear center and rear left/right for example).
- Fixes bug #533817.
+ * 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.
- 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.
+2010-04-30 14:07:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- Adjust unit tests with non-standard 1/2 channel layouts to the more
- correct new behaviour.
+ * configure.ac:
+ * ext/gio/gstgiobasesrc.c:
+ Bump GLib requirement to 2.20
+ See http://gstreamer.freedesktop.org/wiki/ReleasePlanning/GLibRequirement
- Add a unit test for 5.1->Stereo downmixing.
+2010-04-30 13:36:59 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: forward duration query duration during group switch if no cached duration
+ ... such as during first group setup.
+ Fixes #616396.
- * 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.
+2010-04-02 16:37:21 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: only EOS when all streams are EOS
- * 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.
+2010-04-02 16:36:53 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix debug message
- * 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.
+2010-04-30 08:45:43 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-05-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/icles/audio-trickplay.c:
+ test: fix copy and paste error of variable name
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+2010-04-18 20:46:37 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst/adder/gstadder.c:
+ adder: only accept seek-types none and set
+ Previously we were also acting on cur and end, but treating them like none.
-2008-05-28 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-14 23:31:20 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
- Initialise error to NULL as we should.
+ * 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.
-2008-05-28 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-26 17:30:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_query_duration),
- (gst_adder_query_latency), (gst_adder_query):
- Implement latency query.
+ * 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.
-2008-05-27 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-26 11:12:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_query_duration):
- Correctly resync the iterator if gst_iterator_next() returns
- GST_ITERATOR_RESYNC.
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ playbin2: Add flag for enabling/disabling automatic deinterlacing
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-26 11:11:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * win32/vs6/libgstpbutils.dsp:
- Add pbutils-enumtypes.c to sources (#518037).
+ * gst/playback/gstplay-enum.c:
+ playbin: Use g_once_init_{enter,leave} instead of GOnce for enum/flag registration
-2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-23 17:01:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
- * 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.
+2010-04-23 16:24:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/playback/gstplaysink.c:
+ playsink: Correctly reconfigure the video chain when switching from a subtitle to a non-subtitle file
+ Fixes bug #616422.
- * 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).
+2010-04-23 16:08:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: If a text sink is provided, let subtitle parsing be done by decodebin2 if required
+ This way subtitle sinks only get buffers in the format that they
+ understand, i.e. raw parsed text in most cases.
+ Fixes bug #614942.
- * tests/check/pipelines/oggmux.c: (test_pipeline):
- Don't use deprecated function.
+2010-04-23 15:30:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Set subtitle encoding on the decodebins again
- * 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.
+2010-04-23 15:22:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin2.c: (perform_eos):
- When pushing out the EOS event, collect the return value and warn when
- something failed.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: API: Add GST_VIDEO_FORMAT_v308 for packed 4:4:4 YUV
-2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-23 15:14:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for DVCPRO.
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Some random cleanup
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-23 15:06:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
- Change default scaling method from nearest-neighbour to bilinear.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for Y444, Y42B and Y41B
-2008-05-26 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-23 14:42:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/video.c:
- More checks.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Reorder template caps by the amount of information contained in the color formats
-2008-05-25 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-22 15:46:17 -0400 Joshua M. Doe <joshua.doe@us.army.mil>
- * 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/videorate/gstvideorate.c:
+ videorate: add support for video/x-raw-gray
-2008-05-23 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/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.
+ * 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).
-2008-05-22 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-29 15:02:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-overrides.txt:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
+ * configure.ac:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
- * docs/plugins/gst-plugins-base-plugins.interfaces:
- * docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
- * docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggdemux.h:
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggmux.h:
- * gst/audioconvert/audioconvert.c:
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstaudioconvert.h:
- * gst/gdp/gstgdpdepay.h:
- * gst/gdp/gstgdppay.h:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstdecodebin2.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gstplaybin2.c:
- * gst/playback/gsturidecodebin.c:
- * gst/tcp/gstmultifdsink.c:
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.h:
-
- Add some documentation comments, and some new headers to be scanned.
- Rename some internal enum declarations (audioconvert's DitherType and
- NoiseShapingType, GstUnitType from the TCP elements) to match the
- documented GObject type names so that the docs pick them up.
- Name the playbin2 docs markups properly so they get picked up. They'll
- need renaming back when/if playbin2 becomes playbin.
-
- 100% symbol coverage for the plugin docs, booya.
-
-2008-05-22 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
-
- * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
- Fix generation of NV12/NV21 frames. Fixes bug #532454.
-
-2008-05-22 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
-
- * gst/playback/gstdecodebin.c: (remove_fakesink):
- Lock the fakesink before setting the state to NULL and removing it from
- the bin so that a concurrent state change cannot interfere.
- Fixes #534331.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
-
- * docs/Makefile.am:
- Fix installing plugin documentation when gtk-doc is disabled.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Back to development.
- * gst-libs/gst/rtsp/Makefile.am:
- Distribute, don't install md5.h
+=== release 0.10.29 ===
-2008-05-21 Julien Moutte <julien@fluendo.com>
+2010-04-28 02:16:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.29
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-28 01:34:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/vorbis/vorbisdec.c:
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
- Some debug and comment fixes.
-
- * tests/examples/dynamic/addstream.c: (main):
- Fix , to ;
-
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- * gst/playback/decodetest.c: (new_decoded_pad_cb):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
- (cleanup_decodebin):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
- (connect_element), (gst_decode_group_control_demuxer_pad):
- * gst/playback/gstplaybasebin.c: (queue_remove_probe),
- (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
- (mute_group_type):
- * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
- (gst_play_bin_set_property), (handoff), (gen_video_element),
- (gen_text_element), (gen_audio_element), (gen_vis_element),
- (remove_sinks), (add_sink), (setup_sinks):
- * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
- * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
- (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
- (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
- (gen_video_chain), (gen_text_chain), (gen_audio_chain),
- (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
- (gst_play_sink_request_pad):
- * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (cb_newpad):
- * gst/playback/test6.c: (new_decoded_pad_cb):
- * tests/check/elements/audioconvert.c: (GST_START_TEST):
- * tests/check/elements/audiorate.c: (test_injector_chain),
- (do_perfect_stream_test):
- * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/elements/gnomevfssink.c:
- * tests/check/elements/textoverlay.c:
- (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
- * tests/check/elements/videotestsrc.c: (GST_START_TEST):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- * tests/check/pipelines/oggmux.c: (test_pipeline):
- * tests/check/pipelines/streamheader.c: (GST_START_TEST):
- * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
- * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
- * tests/examples/seek/scrubby.c: (make_wav_pipeline):
- * tests/examples/seek/seek.c: (make_mod_pipeline),
- (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
- (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline),
- (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
- (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
- (update_fill), (msg_buffering):
- Don't use bad gst_element_get_pad().
-
-2008-05-21 Stefan Kost <ensonic@users.sf.net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- * gst-libs/gst/riff/riff-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 ';'.
+2010-04-25 23:14:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.28.3 pre-release
- * docs/design/part-playbin2.txt:
- Add some leftover doc.
+2010-04-20 17:20:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-plugins-base.doap:
+ doap: update repository info from cvs->git and maintainers
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
- Fix copy & paste error in last commit.
+2010-04-23 14:39:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From fc85867 to 4d67bd6
- * 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.
+2010-04-22 20:58:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-21 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Fix Y41B->Y444 conversion
+ ...which is the intermediate conversion for conversion to all
+ other formats.
+ Fixes bug #616545.
- Patch by: Henrik Eriksson <henriken at axis dot com>
+2010-04-16 20:03:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
- (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
- (gst_multi_fd_sink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Add support for DSCP QOS. Fixes #469933.
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Don't leak the input buffer in error cases
+ Fixes bug #615572.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-29 12:53:11 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * ext/ogg/gstoggmux.c:
+ docs: fix typo in link name
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-15 12:59:53 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-15 11:44:49 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/audioconvert/gstchannelmix.c:
- (gst_channel_mix_fill_normalize):
- Prevent division by zero if the channel mix matrix contains only
- zeroes.
+ * sys/ximage/ximagesink.c:
+ ximagesink: refactor _update_geometry()
+ Refactor like in xvimagesink. Remove the extra parameter and adjust the assert check.
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-15 07:18:05 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * configure.ac:
+ configure: Drop -Wcast-align
+ Commit message copied from core's commit from Benjamin Otte:
+ 246f5dba96a5b50bb74621af67b30942cca72af5
+ Apparently gcc warns that GstMiniObject is not castable to
+ GstEvent/Message/Buffer due to them containing 64bit variables, even
+ though ARM hackers claim that those only need 4byte alignment. And as
+ long as gcc behaves that way, this warning is not very useful.
+ So we'll remove the warning until this problem is fixed.
+ Fixes #615698
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
- Close a buffer memory leak. Fixes bug #534071.
+2010-04-14 14:13:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- * 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.
+2010-04-14 13:50:21 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-21 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 translations
- * 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.
+2010-04-13 16:20:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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.
+2010-04-14 13:43:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-20 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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.
- * configure.ac:
- Require core CVS for GstBaseSrc buffer caps setting magic.
+2010-04-12 14:02:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add channels and rate to ADTS caps if we can
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Fix logic in last commit.
+2010-04-12 13:33:18 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk>
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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>
- * 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.
+ * 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-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.
+ * 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.
-2008-05-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-26 19:00:47 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_resample_slaving),
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
- (gst_base_audio_sink_async_play),
- (gst_base_audio_sink_change_state):
- Change the way in which the ringbuffer is started when dealing with a
- slaved clock and latency. We now sync to the clock until we reach
- upstream latency before starting the ringbuffer. This has the effect
- that we can accurately align the master and slave clocks and let the
- rate correction code take care of the initial drift or rounding errors
- instead of leaving them uncorrected with the old approach.
-
-2008-05-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: use ptime-multiple
+ Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>
+ Fixes #613248
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_fixate_channels):
- Correctly set the default channel positions when converting to 8
- channels.
+2010-04-09 16:06:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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
- * configure.ac:
- Error out if we don't have the required version of core.
+2010-04-09 11:20:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * common:
+ Automatic update of common submodule
+ From 218568f to fc85867
- * 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.
+2010-04-08 17:49:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-19 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/Makefile.am:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstplayback.h:
+ playback, ogg: dist new gstplayback.h and gstogg.h
- * 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.
+2010-04-09 08:23:33 +0200 Thomas Green <thomasgr33n@googlemail.com>
-2008-05-16 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * gst/adder/gstadder.c:
- Adder doesn't support 24 bit samples so don't claim it supports them
- in the pad template caps.
+2010-04-08 07:39:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Be sure that metadata is writable before setting caps
+ Call gst_buffer_make_metadata_writable before attempting
+ to set caps on the buffer.
- * gst-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.
+2010-04-08 12:21:50 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * ext/gio/gstgio.c:
+ * ext/gnomevfs/gstgnomevfs.c:
+ ext: Invert rank of gio and gnomevfs elements
- * 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.
+2010-04-08 01:26:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-14 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ alsa: don't pass non-constant strings as printf format strings
+ Fixes 'format not a string literal and no format arguments' compiler
+ warning when compiling with -DGST_DISABLE_PRINTF_EXTENSION.
- * gst/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.
+2010-04-07 20:21:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-14 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/video/video.h:
+ docs: add gtk-doc chunks with Since: tags for new GST_VIDEO_CAPS_GRAY* API
- Patch by: Bernard B <b-gnome at largestprime dot net>
+2010-04-07 19:07:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * autogen.sh:
+ * configure.ac:
+ build: bump autoconf requirement to 2.60 for gobject-introspection.m4
+ Require autoconf 2.60 (which was released in June 2006).
+ Fixes #600718.
- * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
- Add a testcase for seqnum compare function.
+2010-04-07 17:25:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-14 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/video/video.c:
+ video: Fix parsing of 8-bit grayscale caps
- * 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().
+2010-04-07 17:21:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-14 Stefan Kost <ensonic@users.sf.net>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/video/video.h:
+ video: API: Add GST_VIDEO_CAPS_GRAY{8,16}
- * sys/xvimage/xvimagesink.c:
- Better debug logging in port value handling. Merging separate port
- value loops into one.
+2010-04-07 17:08:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ video: API: Add gst_video_format_is_gray() to the docs
- Patch by: Hannes Bistry <hannesb at gmx dot de>
+2010-04-07 17:07:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
- * gst/tcp/gsttcpserversink.c:
- (gst_tcp_server_sink_handle_server_read),
- (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
- Fix regression in clientsrc because we did not add the fd to the poll
- set anymore. Fixes #532364.
- Do some cleanups here and there.
+ * win32/common/libgstvideo.def:
+ video: Add new symbol to the exported symbols list
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-07 17:06:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
- Use correct marshallers. GstCaps are a boxed type and no GObject
- subclass.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for 8-bit and 16-bit grayscale formats
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-06 10:55:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * win32/common/libgstrtsp.def:
- Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
- symbols.
+ * 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
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+2010-04-04 21:24:44 -0700 David Schleef <ds@schleef.org>
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+ * configure.ac:
+ configure: fix cdparanoia check
+ Linking with libcdda_paranoia.so requires also linking with
+ libcdda_interface.so.
- * 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.
+2010-04-04 18:00:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-05-13 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/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
- Fix nv12<->nv21 conversion if stride is larger than width.
+2010-04-04 21:18:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-13 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Jan Gerber <j at oil21 dot org>
+2010-04-04 20:55:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: Detect errors from device polling, stop the task and post an error message
+ Partially fixes bug #614545.
-2008-05-12 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-04 12:13:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2008-05-11 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-03 22:49:11 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/pbutils/install-plugins.c:
- Fix docs: type and missing word.
+ * 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.
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-03 14:03:45 +0100 Vincent Untz <vuntz@gnome.org>
- * 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/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.
-2008-05-10 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-04-02 21:01:25 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/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.
-2008-05-09 Jan Schmidt <jan.schmidt@sun.com>
+2010-04-02 20:56:19 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-05-09 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-04-02 18:56:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
- Choose to allocate one less segment but require one additional segment
- as latency.
+ * tests/examples/seek/seek.c:
+ seek: Only use embed_xid if HAVE_X is defined
+ Fixes bug #614622.
- * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
- No need to increment the number of segments in the source.
+2010-04-01 19:13:22 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_get_time), (clock_convert_external),
- (gst_base_audio_sink_resample_slaving),
- (gst_base_audio_sink_skew_slaving),
- (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
- (gst_base_audio_sink_async_play):
- Remove adding latency when returning the internal time while subtracting
- it again when we use the value a little later.
- When calculating the end timestamp, we are making a rounding error
- with the current algorithm. Ensure that we don't accumulate these
- rounding errors when aligning samples by not resampling at all if we
- don't need to. Fixes #419351.
- Make the initial calibration of the clock slaving a little more
- predictable and accurate. Also handle the case where we don't do
- clock slaving.
-
-2008-05-09 Sebastian Dröge <slomo@circular-chaos.org>
-
- Based on a patch by:
- Björn Benderius <bjoern dot benderius at axis dot com>
+ * tests/check/pipelines/basetime.c:
+ tests/basetime: Don't run test with osxaudiosrc
+ libcheck runs the actual tests in a forked process and that makes the guys
+ in Cupertino really sad.
- * gst/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.
+2010-04-01 18:51:17 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-08 Edward Hervey <edward.hervey@collabora.co.uk>
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ tests: Unref the bus once we're done with it
- * 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.
+2010-04-01 16:49:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * common:
+ common: Update for new suppressions
- Patch by: Wouter Cloetens <zombie at e2big dot org>
+2010-04-01 13:55:15 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_create), (md5_digest_to_hex_string),
- (auth_digest_compute_hex_urp), (auth_digest_compute_response),
- (add_auth_header), (gst_rtsp_connection_free),
- (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
- (gst_rtsp_connection_set_auth_param),
- (gst_rtsp_connection_clear_auth_params):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Add Digest authorization support for RTSP connections. See #532065.
+ * gst/playback/gstplaysink.c:
+ gstplaysink: Remove unused variable.
+ The value of klass is never used
- * 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.
+2010-04-01 13:53:37 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-08 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.
- Patch by: Sjoerd Simons <sjoerd at luon dot net>
+2010-04-01 13:51:13 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Let audioresample use the buffer allocation of basetransform instead
- of it's own stuff.
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Remove unused variable
- * 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.
+2010-04-01 13:48:42 +0200 Edward Hervey <bilboed@bilboed.com>
-2008-05-07 Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
+ * ext/gnomevfs/gstgnomevfssink.c:
+ gstgnomevfssink: Return the proper GstFlowReturn.
+ We were always returning GST_FLOW_OK previously even if we encountered errors.
- * 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.
+2010-03-30 23:44:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-07 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/gstbaseaudiosink.c:
- (gst_base_audio_sink_query):
- Report the latency with the new seglatency parameter.
+2010-03-30 20:21:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
- (gst_ring_buffer_acquire):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add new field to the ringbufferspec to specify the expected latency
- between the underlying device read/write pointer, this is needed
- when writing sinks that sit a little closer to the hardware.
- Add some more docs for other fields.
+ * tests/examples/seek/jsseek.c:
+ examples: fix printf format warning in jsseek example
+ Yes, I know about G_GSIZE_FORMAT.
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-30 19:56:56 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_transform_ip):
- Return NOT_NEGOTIATED if we didn't set a process function yet for some
- reason instead of crashing later. Might fix bug #509125.
+ * gst-libs/gst/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.
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-30 19:29:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/Makefile.am:
+ Our RIFF library depends on both the audio and tag libraries
+ Update rules in Makefile.am accordingly.
- * 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.
+2010-03-30 15:10:42 +0200 Robert Swain <robert.swain@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST), (audioconvert_suite):
- Add some unit tests by Tim for checking the NONE channel layouts
- and more than 8 channels and add some more unit tests for channel
- conversions.
+ * gst/playback/gstplaysink.c:
+ playsink: Fix aduio_raw_sink typo
-2008-05-06 Wim Taymans <wim.taymans@collabora.co.uk>
+2009-11-28 21:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin2.c: (connect_pad):
- When autoplugging fails, set the element back to NULL before
- unreffing it.
+ * tests/examples/seek/.gitignore:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/jsseek.c:
+ examples: Add a silly joystick based shuttle example
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-29 20:07:52 -0700 David Schleef <ds@schleef.org>
- * win32/common/libgstaudio.def:
- Add gst_base_audio_src_[sg]et_slave_method() to the exported
- symbols.
+ * ext/theora/gsttheoraenc.c:
+ theoraenc: 0-length packets are delta units
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-29 10:47:31 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_sync),
- (end_sami_element), (characters_sami):
- Remove trailing, leading and double whitespaces.
- Correctly timestamp buffers and output the last buffer too.
+ * 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/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add a simple unit test for SAMI parsing.
+2010-03-29 00:22:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+2010-03-28 21:50:58 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/subparse/samiparse.c: (handle_start_sync),
- (start_sami_element), (end_sami_element), (characters_sami),
- (sami_context_reset):
- Only output characters inside the "sync" elements. There could be
- other elements like "style" that have some content but should
- not be printed. Fixes bug #467911.
+ * gst-libs/gst/Makefile.am:
+ gst-libs: specify dependencies in Makefile.am to make them explicit
-2008-05-05 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-24 09:59:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (set_audio_mute),
- (set_active_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (playbin_set_audio_mute):
- Allow setting -1 as current-audio to mute the current audio stream,
- similar to what is done for subtitles. Fixes bug #342294.
+ * gst-libs/gst/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>
-2008-05-05 Edward Hervey <edward.hervey at collabora co uk>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: we can handle avi in download mode too
+ Add avi to the whitelisted types that can be used for download buffering.
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- It's SorensOn and not SorensEn.
+2010-03-26 15:57:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-04 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: Provide packet duration function for old FLAC mapping too
+ Fixes bug #613809.
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Fix description of video/x-flash-video.
+2010-03-18 22:12:40 +0000 Damien Lespiau <damien.lespiau@intel.com>
-2008-05-04 Sebastian Dröge <slomo@circular-chaos.org>
+ * autogen.sh:
+ autogen.sh: Don't call configure with --enable-plugin-docs
+ configure gives a nice warning:
+ configure: WARNING: unrecognized options: --enable-plugin-docs
+ and indeed, I could not find anything in the configure.ac or the m4
+ macros that would allow enabling that option. Remove it then.
- * gst-libs/gst/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.
+2010-03-24 23:04:43 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/audioconvert/gstaudioquantize.c:
- (gst_audio_quantize_free_noise_shaping):
- Don't return before freeing the noise shaping history.
+ * 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
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-24 14:43:21 -0300 Thiago Santos <thiago.sousa.santos@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-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.
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-24 10:18:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (handle_buffer),
- (gst_sub_parse_sink_event):
- * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
- (tmplayer_parse_line):
- Fix parsing of tmplayer subtitle variant where every single line contains
- text and there isn't an empty line after each line to determine the
- duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
- making sure that we push out the last line of text without a duration if
- there's still text left in the buffer at the end.
+ * gst-libs/gst/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
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+2010-03-23 09:32:40 -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:
+ * 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>
-2008-05-03 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * 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/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).
+2010-03-24 12:10:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-05-02 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
- * 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.
+2010-03-23 18:36:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-02 Wim Taymans <wim.taymans@collabora.co.uk>
+ * win32/common/libgstrtsp.def:
+ win32: Add new gst_rtsp_lower_trans_get_type() symbol to the symbol lists
- * 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.
+2010-03-23 11:01:17 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-05-02 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add some more fourcc for MPEG-4 video
- * 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 :)
+2010-03-22 09:15:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-05-02 Thijs Vermeir <thijsvermeir@gmail.com>
+ * configure.ac:
+ configure: require core git
- * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
- (paint_setup_NV21), (paint_hline_NV12_NV21):
- Add support for NV12 and NV21 in videotestsrc
+2010-03-22 08:38:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-05-02 Sebastian Dröge <slomo@circular-chaos.org>
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft.pc.in:
+ pkgconfig: Add @LIBM@ to the FFT pkg-config files
- * 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.
+2010-03-22 08:35:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-audio-uninstalled.pc.in:
+ * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
+ * pkgconfig/gstreamer-fft-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
+ * pkgconfig/gstreamer-floatcast.pc.in:
+ * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
+ * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
+ * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-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/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.
+2010-03-20 13:42:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-30 Michael Smith <msmith@songbirdnest.com>
+ * 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
- * ext/theora/theoradec.c:
- Cool kids don't divide by zero.
- Treat PAR of x:0 as 1:1.
- Fixes #530719.
+2010-03-20 12:55:36 +0000 Torsten Schönfeld <kaffeetisch@gmx.de>
-2008-04-30 Tim-Philipp Müller <tim.muller at collabora co uk>
+ * gst-libs/gst/interfaces/xoverlay.c:
+ docs: add Since: tags to gst_x_overlay_handle_event() docs
+ Fixes #613403.
- * 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.
+2010-03-19 22:33:58 +0100 Benjamin Otte <otte@redhat.com>
-2008-04-28 Michael Smith <msmith@songbirdnest.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Constify some strings in the API
+ Needed by plugins-good
- * 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.
+2010-03-19 16:41:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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()
+2010-03-19 15:37:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Fix the docs about the seqnum compare function, it returns a difference.
+2010-03-19 15:00:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-04-24 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtsptransport.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: add GType for transport flags
+ Make a method to register the transport flags as a GType.
- * ext/alsa/gstalsadeviceprobe.c:
- (gst_alsa_get_device_list): Don't return before freeing up
- the allocated structures.
+2010-03-19 01:00:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-24 Stefan Kost <ensonic@users.sf.net>
+ * ext/cdparanoia/Makefile.am:
+ * ext/gio/Makefile.am:
+ * ext/gnomevfs/Makefile.am:
+ * ext/libvisual/Makefile.am:
+ * ext/ogg/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ * gst/ffmpegcolorspace/Makefile.am:
+ * gst/tcp/Makefile.am:
+ * gst/videotestsrc/Makefile.am:
+ * sys/v4l/Makefile.am:
+ * tests/examples/app/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tests/icles/Makefile.am:
+ build: Makefile.am fixes
+ Mostly just add missing $(GST_BASE_CFLAGS), but also fix up order
+ of flags (see docs/random/moving-plugins).
- * gst/playback/gstplaybin.c:
- Remove obsolete streaminfo code and fix a leak. Fixes #529546
+2010-03-19 00:46:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore new unit test binary
- * ext/ogg/gstoggdemux.c:
- Revert the event part, that should not go in.
+2010-03-17 23:57:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ configure.ac: -Wmissing-prototypes and -Wnested-externs are not valid for C++
+ Fixes building Qt-based overlay examples in combination with -Werror.
- * ext/ogg/gstoggdemux.c:
- Don't leak GstPluginFeatures when filtering.
+2010-03-17 16:32:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-23 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ configure.ac: wrap overly long warning flag lines
- * sys/xvimage/xvimagesink.c:
- Add some logging for cases when grabbing the xv failed.
+2010-03-17 19:24:27 -0300 Reuben Dowle <reube.dowle@navico.com>
-2008-04-21 David Schleef <ds@schleef.org>
+ * sys/ximage/ximagesink.c:
+ ximagesink: Fix caps leak
+ Unref caps when peer doesn't accept caps
+ Fixes #613198
- * 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.
+2010-03-17 08:13:59 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-04-21 David Schleef <ds@schleef.org>
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/capsfilter-renegotiation.c:
+ tests: capsfilter-renegotiation: Adds a new unit test
+ Adds a new test for checking that capsfilter 'caps' property
+ changes cause caps renegotiation on the pipeline.
- * sys/xvimage/xvimagesink.c:
- Fix typo that causes the overlay keying color to bright green
- on a 16-bit display. Dark grey good. Bright green bad.
+2010-03-17 16:46:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-21 Stefan Kost <ensonic@users.sf.net>
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Use correct boundary checks for YUY2/UYVY
+ Fixes bug #613093.
- * ext/gnomevfs/gstgnomevfsuri.c:
- Add FIXME comment about using uri-list for source and sink.
+2010-03-17 16:39:13 +0100 Peter Kjellerstedt <peter.kjellerstedt@axis.com>
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * 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.
+2010-03-17 15:41:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-20 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/volume/gstvolume.c:
+ volume: Remove useless cast
+ It's not necessary anymore after latest core change to GstValueArray.
- * 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.
+2010-03-17 12:08:30 +0100 Benjamin Otte <otte@redhat.com>
+
+ * 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.
- Add locking to the properties and use
- gst_element_class_set_details_simple() instead of a static
- GstElementDetails struct.
+2010-03-17 11:14:29 +0100 Benjamin Otte <otte@redhat.com>
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
- (plugin_init):
- Add "mpp" and "mp+" as possible extensions for MusePack files.
+2010-03-17 10:53:21 +0100 Benjamin Otte <otte@redhat.com>
- Add typefinding for MusePack StreamVersion 8 files and include the
- stream version in the caps.
+ * 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.
-2008-04-19 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-17 10:51:57 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/rtp/gstrtppayloads.c:
- (gst_rtp_payload_info_for_name):
- Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
+ * 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
-2008-04-18 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-17 10:47:07 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * 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.
-2008-04-18 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-03-16 20:05:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Don't validate the payload if there isn't any.
- Fixes #525915
+ * 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
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-16 18:31:15 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
- Use g_atomic_int_set() instead of gst_atomic_int_set().
+ * ext/alsa/gstalsaplugin.c:
+ * ext/ogg/gstoggdemux.c:
+ More ENABLE_NLS fixes
-2008-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-16 18:06:16 +0100 Benjamin Otte <otte@redhat.com>
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Return NULL instead of a gchar * array with one NULL element if we
- don't get any supported URI schemes from GIO.
+ * gst-libs/gst/gettext.h:
+ Fix for ENABLE_NLS being undefined for -Wundef
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-03-15 22:49:53 +0100 Benjamin Otte <otte@redhat.com>
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Remove cpp style commented old code.
+ * configure.ac:
+ * ext/libvisual/visual.c:
+ * ext/theora/gsttheoraenc.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ * gst-libs/gst/tag/tags.c:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ * gst/volume/gstvolume.c:
+ * sys/v4l/gstv4lelement.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/gdpdepay.c:
+ * tests/check/elements/playbin.c:
+ * tests/check/elements/playbin2.c:
+ * tests/check/elements/videorate.c:
+ * tests/check/libs/pbutils.c:
+ * tests/check/libs/video.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ * tests/icles/stress-playbin.c:
+ Add -Wwrite-strings to configure
+ Fixes for the code included
-2008-04-15 Stefan Kost <ensonic@users.sf.net>
+2010-03-16 15:45:23 +0100 Benjamin Otte <otte@redhat.com>
+ * 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:
- Fix signal docs.
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gststreamselector.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/subparse/gstssaparse.c:
+ * gst/subparse/gstsubparse.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * sys/v4l/gstv4ljpegsrc.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * sys/v4l/gstv4lsrc.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/playbin.c:
+ * tests/check/elements/playbin2.c:
+ * tests/check/elements/textoverlay.c:
+ * tests/check/libs/cddabasesrc.c:
+ * tests/check/libs/pbutils.c:
+ * tests/old/testsuite/alsa/formats.c:
+ * tests/old/testsuite/alsa/sinesrc.c:
+ gst_element_class_set_details => gst_element_class_set_details_simple
+ Also change my email from the old university one to the current one.
-2008-04-14 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 22:17:56 +0100 Benjamin Otte <otte@redhat.com>
- * 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.
+ * configure.ac:
+ Add -Wundef flag
-2008-04-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-16 16:15:39 +0100 Wim Taymans <wim.taymans@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.
+ * 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/libs/video.c:
- Fix unit test not to create caps with width=0 and height=0.
-
-2008-04-11 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/design/draft-keyframe-force.txt:
- Fix typo.
+2010-03-11 14:52:09 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstqueue2.c: (update_buffering),
- (gst_queue_handle_src_query):
- Set buffering mode in the messages.
- Set buffering percent in the query.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
+ Fixes #612223.
- * 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.
+2010-03-16 01:08:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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.
+2010-03-15 12:54:32 -0500 Rob Clark <rob@ti.com>
-2008-04-09 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP7 fourccs
+ Fixes #612968.
- * 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.
+2010-03-15 12:54:01 -0500 Rob Clark <rob@ti.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for On2 VP62 fourcc
+ See #612968.
-2008-04-09 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-15 23:46:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/multichannel.c:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/interfaces/tuner.c:
+ * gst-libs/gst/pbutils/install-plugins.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ docs: more helper libraries docs fixes
+ Quieten gtk-doc a bit more.
-2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-15 23:47:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/rtsp/gstrtspextension.c:
+ docs: add GstRTSPExtension to docs
+ Add minimal docs for GstRTSPExtension so people know it exists.
-2008-04-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 18:45:13 +0000 David Hoyt <dhoyt@llnl.gov>
- * NEWS:
- * RELEASE:
- * gst-plugins-base.doap:
- Merge other changes from 0.10.19 release branch.
+ * 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.
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 13:39:58 +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).
+ * gst-libs/gst/tag/gstxmptag.c:
+ docs: fix typo in gst_tag_list_from_xmp_buffer() docs chunk
-2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-15 13:32:58 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- Remove some more fields.
+ * 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
-2008-04-06 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-15 13:24:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: add new libgstvideo API to documentation
- * configure.ac:
- Actually build dlls when cross-compiling with mingw32.
- Fixes bug #526247.
+2010-03-15 13:19:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-03 Tim-Philipp Müller <tim at centricular 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()
- * configure.ac:
- Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
+2010-03-15 13:14:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videotestsrc/videotestsrc.c:
+ * gst/videotestsrc/videotestsrc.h:
+ videotestsrc: use C comments instead of C++-style comments
- * 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.
+2010-03-15 13:10:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: use g_value_set_static_string() for string constants
- * ext/ogg/gstoggmux.c:
- Fix sample pipeline description.
+2010-03-15 14:26:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-04-03 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaysink.c:
+ playsink: Avoid g_object_set() on NULL if a text sink is used
+ Fixes bug #611702.
- * 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
+2010-03-15 14:10:09 +0100 Sebastian Dröge <sebastian.droege@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-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.
+ * gst/subparse/gstsubparse.c:
+ subparse: Correctly escape brackets in DKS regex
+ Fixes bug #612783.
- * ext/ogg/gstoggmux.c:
- Document oggmux.
+2010-03-15 11:36:22 +0100 Wim Taymans <wim.taymans@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.
+ * 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.
-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>
+2010-03-15 11:41:35 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 01:35:15 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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.
-2008-04-03 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 01:34:09 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * ext/theora/gsttheoradec.c:
+ theoradec: add chroma-site to caps
-2008-04-03 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-15 01:33:36 -0700 David Schleef <ds@schleef.org>
- * gst-libs/gst/cdda/Makefile.am:
- * gst-libs/gst/cdda/base64.c:
- * gst-libs/gst/cdda/base64.h:
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cddabasesrc_calculate_musicbrainz_discid):
- Use GLib's base64 implementation instead of our own.
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: add chroma-site to caps
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-15 01:31:20 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: add gst_video_parse_caps_chroma_site()
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-14 19:10:16 -0700 David Schleef <ds@schleef.org>
- Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
+ * ext/theora/gsttheoradec.c:
+ theoradec: add color-matrix to caps
- * 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.
+2010-03-14 16:17:46 -0700 David Schleef <ds@schleef.org>
-2008-04-02 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add color-matrix to template caps
- * 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...
+2010-03-14 22:14:19 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-02 Wim Taymans <wim.taymans@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/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.
+2010-03-14 22:13:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-04-01 Sebastian Dröge <slomo@circular-chaos.org>
+ * Makefile.am:
+ build: add cruft alert for common/shave* leftovers to top-level Makefile.am
- * configure.ac:
- Require GLib 2.12 and liboil 0.3.14.
+2010-03-14 13:11:53 -0700 David Schleef <ds@schleef.org>
- * gst/volume/gstvolume.c: (volume_process_double):
- Unconditionally use liboil 0.3.14 function.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Don't drop zero-sized packets
+ Zero-sized packets have relevence to Theora.
-2008-03-31 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-12 15:47:40 +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/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.
-2008-03-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-12 15:44:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- MP4S is generic MPEG-4, not a microsoft variant.
+ * gst/volume/gstvolume.c:
+ volume: Only allocate a mute value array if a control source exists for the mute property
-2008-03-27 Michael Smith <msmith@fluendo.com>
+2010-03-12 13:55:55 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
- Check the body CRC (if set) when depayloading.
- Fixes #522401.
+ * common:
+ Automatic update of common submodule
+ From e272f71 to 55cd514
-2008-03-24 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 10:50:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
- Fix Since: version for new property.
+ * gst-libs/gst/tag/gstxmptag.c:
+ tags: Add new mapping to XMP helpers
+ Adds geotagging mappings to XMP helpers
+ Fixes #609539
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 20:16:44 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
- (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
- Don't error when poll_wait returns EAGAIN.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstqueue2.c: (gst_queue_is_filled):
- The queue is never filled when there are no buffers in the queue at all.
- Fixes #523993.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (init_group), (free_group), (gst_play_bin_init),
- (gst_play_bin_finalize), (gst_play_bin_set_uri),
- (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
- (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
- (gst_play_bin_set_current_video_stream),
- (gst_play_bin_set_current_audio_stream),
- (gst_play_bin_set_current_text_stream),
- (gst_play_bin_set_encoding), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
- (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
- (activate_group), (deactivate_group), (setup_next_source),
- (save_current_group), (gst_play_bin_change_state):
- Update some docs.
- Add new locks and conds to protect pipeline creation and group
- switching.
- Implement the sub-uri property.
- Keep track of pending uridecodebin creation and configure the output
- pipeline after all streams are configured.
- Propagate subtitle encoding to the uridecodebins.
- Implement getting the video/audio/visualisation elements.
- Use input-selector for stream switching.
- If we are asked to do visualisation, prefer to autoplug raw sinks
- instead of sinks that accept encoded data.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
- (gst_play_sink_init), (gst_play_sink_dispose),
- (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
- (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
- (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
- (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
- (gst_play_sink_set_volume), (gst_play_sink_get_volume),
- (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
- (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
- (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
- (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Add methods to get audio/video/vis elements.
- Add methods to set the font description for the overlay.
- Remove properties, we're using this element with its methods only.
- Add support for subtitles.
- Rearrange the locking a bit to not use the object lock for protecting
- the pipeline construction.
- Try to use the volume and mute property on the sink when its available.
- Implement the mute option with volume when the sink does not have a mute
- property.
- Only add volume element when the sink has no volume property.
- Only do visualisations with raw audio pads.
-
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
- (gst_text_overlay_init), (gst_text_overlay_set_property),
- (gst_text_overlay_get_property), (gst_text_overlay_src_event),
- (gst_text_overlay_text_event), (gst_text_overlay_video_event),
- (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
- (gst_text_overlay_change_state):
- * ext/pango/gsttextoverlay.h:
- Add property to configure waiting for text on the textpad or not, with
- the default behaviour being the old one (always wait for text before
- rendering the video). This default behaviour is usually not the best one
- because the text stream can very sparse and could require queueing a lot
- of video.
- Fix the flushing and EOS handing so that we don't mix up their meaning.
+ * gst-libs/gst/interfaces/Makefile.am:
+ Don't have 2 include dirs
+ Seems to have been accidentally introduced in
+ 7269bc26d0a4bf44bd77a039fb54777625ef5f39.
-2008-03-24 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 16:35:10 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * tests/icles/audio-trickplay.c:
+ tests: Fix another unitialized variable
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Fix bug in determining the sample start/stop position, we want to base
- this decision on the fact that we are going forwards or backwards, not
- slower or faster. This fixes some ugly resync warnings when playing at
- very slow speeds.
+2010-03-11 16:09:26 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-03-23 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/icles/audio-trickplay.c:
+ tests: Fix unitialized variable.
- * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
- Correctly set the supported URI schemes and don't leave
- some schemes in the middle or at the start at NULL.
+2010-03-11 15:38:18 +0100 Benjamin Otte <otte@redhat.com>
-2008-03-23 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ * ext/ogg/gstoggdemux.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/vorbis/gstvorbistag.c:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioquantize.h:
+ * gst/audioconvert/gstchannelmix.h:
+ * gst/playback/gstplaysink.c:
+ Add -Wredundant-decls to warning flags
+ ... and fix all the warnings that flag throws.
- * tests/check/elements/gdpdepay.c:
- Make test compile without unused function/variable warnings on PPC.
-
-2008-03-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-11 13:32:14 +0100 Benjamin Otte <otte@redhat.com>
* 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>
+ * 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.
- * configure.ac:
- Fix lrint/lrintf checks to actually work. These functions are
- in libm on Linux at least so try to link to it.
+2010-03-11 12:49:02 +0100 Benjamin Otte <otte@redhat.com>
-2008-03-21 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/gio/gstgiobasesink.c:
+ gio: Remove unused function
- * configure.ac:
- Back to development - 0.10.18.1
+2010-03-11 11:14:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.10.18 ===
+ * ext/vorbis/gstvorbisparse.c:
+ vorbisparse: make sure header buffer metadata is writable before modifying it
+ Fixes unit test failures with core git.
-2008-03-20 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-11 12:18:00 +0100 Benjamin Otte <otte@redhat.com>
- * configure.ac:
- releasing 0.10.18, "I will follow"
+ * 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.
-2008-03-18 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-11 12:04:32 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
- * win32/common/config.h:
- 0.10.17.4 pre-release
+ Add WARNING_CXXFLAGS where ERROR_CXXFLAGS are
+ This matches the previous commit doing the same for CFLAGS in response
+ to the common/ module changes.
-2008-03-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 12:04:37 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * 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
-2008-03-18 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-11 10:38:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * 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.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
- (read_line), (gst_rtsp_connection_read_internal):
- Generic Windows fixes that makes libgstrtsp work on Windows when
- coupled with the new GstPoll API. See #520808.
+2010-03-11 11:20:48 +0100 Benjamin Otte <otte@redhat.com>
-2008-03-17 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From df8a7c8 to e272f71
- Patch by: Milosz Derezynski <internalerror at gmail dot com>
+2010-03-11 10:55:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
- If seeking to a new position succeeds don't simply return from
- create() without creating a buffer. Do this only in the case
- seeking to the new position fails. Fixes bug #523054.
+ * gst-libs/gst/interfaces/xoverlay.c:
+ xvoverlay: correct version number in docs
-2008-03-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-26 13:56:21 +0200 Stefan Kost <ensonic@users.sf.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).
+ * 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.
- * 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.
+2010-02-08 17:20:35 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-14 David Schleef <ds@schleef.org>
+ * 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.
- * gst/videotestsrc/videotestsrc.c: Oops, revert last change
- because -base is in freeze.
+2010-02-19 14:38:36 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-03-14 David Schleef <ds@schleef.org>
+ * 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.
- Patch by: William M. Brack
+2010-02-19 14:46:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
+ * sys/xvimage/xvimagesink.c:
+ * sys/xvimage/xvimagesink.h:
+ xvimagesink: implement set_render_rectangle
+ Previously we hardcoded the target rectangle passes to Xv(Shm)PutImage. Extend
+ the implementation to use a full rectangle and don't assume 0,0 for top,left.
-2008-03-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-17 15:00:13 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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()
-2008-03-12 Michael Smith <msmith@fluendo.com>
+2010-02-16 12:06:08 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: take new size from event thread and do not poll for every frame
+ We can update the geometry in ConfigureNotify (unless we disable event-
+ handling). If event handling is disabled, one should use _expose() to trigger a
+ redraw and update the geometry.
-2008-03-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-10 21:51:59 +0100 Benjamin Otte <otte@redhat.com>
- * win32/common/libgstnetbuffer.def:
- Add new symbol gst_netaddress_equal. Fixes bug #521743.
+ * common:
+ Automatic update of common submodule
+ From 9720a7d to df8a7c8
-2008-03-11 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-10 21:01:20 +0100 Benjamin Otte <otte@redhat.com>
* configure.ac:
- * win32/common/config.h:
- 0.10.17.3 pre-release
+ 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.
-2008-03-10 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-10 20:43:46 +0100 Benjamin Otte <otte@redhat.com>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Fix duration when no clock was provided. Fixes #520300.
-
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
-
- 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
+ * common:
+ Automatic update of common submodule
+ From 0b6e072 to 9720a7d
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-10 16:09:45 +0100 Benjamin Otte <otte@redhat.com>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Update mode property docs, it's deprecated now.
+ * common:
+ Automatic update of common submodule
+ From 7cc5eb4 to 0b6e072
-2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-10 14:36:34 +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.
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: do not cache FLUSH_START/_STOP events
+ ... and similarly so for serialized events.
-2008-03-04 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-10 14:34:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- * win32/common/config.h:
- 0.10.17.2 pre-release
+ * gst/playback/gstplaysink.c:
+ playsink: provide correct error message if configured audio/video sink fails
-2008-03-03 Jan Schmidt <jan.schmidt@sun.com>
+2010-03-10 10:22:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/Makefile.am:
- GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
- them twice
+ * ext/vorbis/gstvorbisdec.h:
+ vorbisdec: remove unused field
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Add new API to the defs
+2010-02-02 11:34:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2008-03-03 Peter Kjellerstedt <pkj@axis.com>
+ * tests/check/pipelines/vorbisdec.c:
+ tests: enable strict discontinuity checking on vorbisdec pipeline
+ Closes #423086.
- Patch by: Mersad Jelacic <mersad at axis dot com>
+2010-03-10 01:09:31 +0000 Tim-Philipp Müller <tim.muller@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)
+ * common:
+ Automatic update of common submodule
+ From 7aa65b5 to 7cc5eb4
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 01:07:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * tests/check/libs/mixer.c:
- Add a few simple checks for the new message types.
+ * gst-libs/gst/video/video.c:
+ docs: fix Returns: for gst_video_parse_caps_color_matrix()
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-10 00:46:34 +0000 Tim-Philipp Müller <tim.muller@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.
-
-2008-03-03 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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)
-
-2008-03-03 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. (#498222)
-
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for IMelody files, using audio/x-imelody.
- See bug #519516.
-
-2008-03-03 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update for changed string
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Don't include unix headers unconditionally (fixes #518037).
+2010-03-10 00:42:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
- (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
- (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
- (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
- (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
- (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
- (gst_video_format_is_packed), (video_format_is_packed):
- Add unit test that makes sure that the strides, offsets and
- sizes returned for the various YUV formats by the new video API
- match the old reference implementation in videotestsrc.
+2010-03-10 00:41:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/videorate.c:
+ tests: don't use deprecated functions in videorate unit test
- * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
- (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
- (gst_video_format_is_rgb), (gst_video_format_is_yuv),
- (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
- (gst_video_format_get_pixel_stride),
- (gst_video_format_get_component_width),
- (gst_video_format_get_component_height),
- (gst_video_format_get_component_offset), (gst_video_format_get_size):
- * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
- (GST_VIDEO_FORMAT_Y42B):
- API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
+2010-03-10 00:29:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-03-02 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstvideo.def:
+ win32: add new API to libgstvideo.def
- * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
- YV12 is I420 with swapped components 1 and 2, so the offset of
- component 1 for I420 should be the offset for component 2 for YV12
- and vice versa.
+2010-03-09 15:39:55 -0800 David Schleef <ds@schleef.org>
-2008-02-29 Rene Stadler <mail@renestadler.de>
+ * ext/ogg/gstoggmux.c:
+ oggmux: Don't flush after every frame for theora
- * sys/v4l/gstv4lelement.c:
- Add missing semicolon to fix indentation.
+2010-03-09 21:26:58 +0000 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-29 Julien Moutte <julien@fluendo.com>
+ * common:
+ Automatic update of common submodule
+ From 44ecce7 to 7aa65b5
- * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
- (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to detect
- if we can do SPDIF output.
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
- (gst_alsasink_prepare), (gst_alsasink_close), (gst_alsasink_write):
- * ext/alsa/gstalsasink.h: Initial support for SPDIF.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
- * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer types
- to support AC3, EC3 and IEC958 buffers.
+2010-03-09 13:05:23 -0800 David Schleef <ds@schleef.org>
-2008-02-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add color-matrix handling to caps
- * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
- (gst_mixer_message_parse_mute_toggled),
- (gst_mixer_message_parse_record_toggled),
- (gst_mixer_message_parse_volume_changed),
- (gst_mixer_message_parse_option_changed):
- De-cruft and fix message type assertions (NULL is not a really
- valid mixer message type string).
+2010-01-30 22:55:01 -0800 David Schleef <ds@schleef.org>
-2008-02-29 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: Add color-matrix to caps
- * ext/libvisual/visual.c: (gst_vis_src_negotiate):
- When negotiating, actually start from a format that we can support
- instead of from the too generic template.
+2010-02-26 16:25:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-29 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:
+ * pkgconfig/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tools/Makefile.am:
+ build: Make some more rules silent if requested
- * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
- Enable vis setting.
+2010-02-26 15:40:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaysink.c: (gst_play_sink_init),
- (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
- (gen_vis_chain):
- Implement vis switching while playing.
+ * 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.
-2008-02-28 David Schleef <ds@schleef.org>
+2010-02-23 19:12:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
+ * 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.
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-17 19:18:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ volume: If a controller is used, use sample accurate property values
+ Fixes bug #609801.
- * gst/tcp/Makefile.am:
- * gst/tcp/fdsetstress.c:
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstfdset.h:
- Removed fdset and stress test, they are now known as GstPoll in
- core.
-
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
- (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_queue_buffer),
- (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
- (gst_multi_fd_sink_stop):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
- (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
- (gst_tcp_gdp_read_caps):
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
- (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
- (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
- (gst_tcp_client_src_create), (gst_tcp_client_src_start),
- (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
- (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
- (gst_tcp_server_src_create), (gst_tcp_server_src_start),
- (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
- * gst/tcp/gsttcpserversrc.h:
- Port to GstPoll. See #505417.
+2010-03-09 19:17:04 +0100 Benjamin Otte <otte@redhat.com>
-2008-02-28 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/video/video.c:
+ gstvideo: Fix typos in comments
- Patch by: Peter Kjellerstedt <pkj at axis com>
+2010-03-09 17:32:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
- (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
- (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
- (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
- (gst_rtsp_connection_flush):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- Use GstPoll for the rtsp connection. See #505417.
+ * 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
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+=== release 0.10.28 ===
- * 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.
+2010-03-08 23:20:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-27 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.28
- * tests/check/libs/rtp.c: (GST_START_TEST):
- Add check for RTP buffer defaults, padding and marker bit API.
+2010-03-08 23:19:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-27 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:
+ Update .po files
- * 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.
+2010-03-08 21:57:03 +0100 Benjamin Otte <otte@redhat.com>
-2008-02-27 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoraenc.c:
+ theora: Fix SIGFPE when using 0/1 framerate
+ libtheora crashes with a 0 framerate, so let's forbid it.
+ https://bugzilla.redhat.com/show_bug.cgi?id=571289
- * gst-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().
+2010-03-08 14:50:25 +0000 David Schleef <ds@schleef.org>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * ext/ogg/dirac_parse.c:
+ oggdemux: fix dirac header parsing
+ Fixes #611900.
- * 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.
+2010-03-08 14:46:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * tests/examples/overlay/Makefile.am:
+ examples: make sure to dist qtgv-xoverlay.h header file
+ This time for real.
+ Fixes #610832.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for the smoke video codec. Copied from the jpeg plugin.
+2010-03-08 12:11:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-25 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basedepay: clarify some documentation
- * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
- (plugin_init):
- Add midi typefinder, copied from the timidity plugin.
+2010-03-08 11:25:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/alsa/gstalsasrc.c:
+ alsasrc: return right number of bytes that we wrote
- Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
+2010-03-08 11:20:51 +0100 Dake Gu <gudake@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).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: fix handling of x-server-ip-address
+ Fix handling of x-server-ip-address.
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-02 11:25:31 +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.
+ * docs/design/draft-keyframe-force.txt:
+ docs: update keyframe force event
+ Add field to send all headers.
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
+=== release 0.10.27 ===
- Patch by: Brian Cameron <brian dot cameron at sun dot com>
+2010-03-06 00:09:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- Don't hardcode -Wall and -Werror for configure checks, this fails
- with non-GCC compilers. Fixes bug #517991.
-
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
-
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audioconvert/gstaudioquantize.c:
- * gst/audioconvert/gstaudioquantize.h:
- Make audioconvert gap aware. If noiseshaping is used, wait for
- noiseshaping to converge before marking as GAP. Fixes #517813.
-
-2008-02-21 Stefan Kost <ensonic@users.sf.net>
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.27
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
+2010-03-06 00:08:23 +0000 Tim-Philipp Müller <tim.muller@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:
+ Update .po files
-2008-02-20 Sebastian Dröge <slomo@circular-chaos.org>
+2010-03-05 15:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/gnomevfs/gstgnomevfssink.c:
- (gst_gnome_vfs_sink_handle_event):
- Return FALSE when seeking for a new segment fails instead
- of silently ignoring the failure and appending every buffer
- that comes for the new segment.
+ * configure.ac:
+ 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.
+
+2010-03-03 20:18:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.4 pre-release
-2008-02-20 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-03-03 20:17:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * po/ja.po:
+ po: update translations
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-03 20:15:44 +0000 Josep Torra Valles <n770galaxy@gmail.com>
- * 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'.
+ * 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.
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-03-01 12:17:40 +0100 Wim Taymans <wim.taymans@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.
+ * 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
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-24 01:10:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/multichannel.c:
- Fix typo in docs blurb.
+ * configure.ac:
+ * po/nl.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.3 pre-release
-2008-02-19 Julien Moutte <julien@fluendo.com>
+2010-02-23 16:57:53 +0100 Götz Waschk <waschk@mandriva.org>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * tests/examples/overlay/Makefile.am:
+ examples: Dist header file for the Qt graphics view example
+ Fixes bug #610832.
- * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
- typefind lookup to fix typefinding on HD clips.
+2010-02-23 11:41:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-02-19 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: use the chain begin_time instead of our counter
+ We update the passed begintime argument to narrow our search region in the
+ binary search. This means that it does not always contain the chain begin time
+ after a couple of bisects. Use the real chain->begin_time to bring the
+ granuletime to the time in the chain instead.
+ Fixes #610005
- * 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).
+2010-02-19 18:24:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2008-02-19 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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
- * 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.
+2010-01-27 15:07:47 -0300 Thiago Santos <thiago.sousa.santos@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/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
+
+2010-02-21 19:52:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
- Use last-buffer property of the video sink to get a video snapshot.
+ * 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.
- * 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.
+2010-02-22 13:01:19 +0200 Stefan Kost <ensonic@users.sf.net>
-2008-02-19 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- Patch by: Josep Torra Valles <josep at fluendo dot com>
+2010-02-19 14:40:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
- (plugin_init):
- Add typefinding support for h264 elementary streams.
- Fixes bug #517420.
+ * tests/examples/overlay/.gitignore:
+ gitignore: ignore files in new example directroy
-2008-02-18 Stefan Kost <ensonic@users.sf.net>
+2010-02-17 14:59:33 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Require CVS of core for new API in collectpads.
+ * gst-libs/gst/video/Makefile.am:
+ make: fix copy and paste error in git rules (audio<->video)
- * gst/adder/gstadder.c:
- Use new API to make adder sparse stream aware.
+2010-02-19 17:44:18 +0100 Sebastian Dröge <sebastian.droege@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/gstplaysink.c:
+ playsink: Ghost the video sinkpad if a text sinkpad is available
+ Only don't ghost it if no visualizations are need and if
+ no text is needed and no textchain was created yet.
+ Fixes bug #610379.
- * gst/playback/gststreamselector.c: (gst_selector_pad_event),
- (gst_selector_pad_chain):
- * gst/playback/gststreamselector.h:
- Make sure we forward the event only once.
+2010-02-19 00:22:13 +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.
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.26.2 pre-release
-2008-02-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-19 00:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: update translation files
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
- Add some more debug info.
- Make sure we never return a negative delay. Fixes #516246.
+2010-02-19 00:17:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/overlay/.gitignore:
+ Ignore new overlay examples
- * 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.
+2010-02-18 23:47:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-02-12 Julien Moutte <julien@fluendo.com>
+ * tests/examples/overlay/gtk-xoverlay.c:
+ examples: don't hard-code xvimagesink for Gtk+ GstXOverlay example
+ Try to find a working videosink, don't hardcode xvimagesink. Also
+ add some borders to window and give it a title so that it's clear
+ that this is really a Gtk+ window and not a window created by the
+ videosink.
- * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
- some seek flags when changing rate.
+2010-02-18 11:42:55 -0800 David Schleef <ds@schleef.org>
-2008-02-12 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/tcp/gsttcp.c:
+ tcp(client/server)src: Fix handling of closed sockets
+ The peer closing the socket should cause an EOS, instead of
+ silently doing nothing. This changes the behavior to be
+ more like fdsrc. Fixes: #610386
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer):
- Fix potential leaks.
+2010-02-18 12:42:53 +0000 Patrick Radizi <patrick.radizi@axis.com>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
- Fix leak when there is no function configured.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: make sure not to dereference NULL username or password
+ Fixes #610268.
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-17 21:22:54 -0800 David Schleef <ds@schleef.org>
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
- (gst_v4lsrc_buffer_finalize):
- Correctly chain up the finalize method.
+ * 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.
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-16 15:43:26 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/gio/gstgiostreamsink.c:
- * ext/gio/gstgiostreamsrc.c:
- Add documentation and example code for giostreamsink/giostreamsrc.
+ * configure.ac:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/overlay/Makefile.am:
+ * tests/examples/overlay/gtk-xoverlay.c:
+ * tests/examples/overlay/qt-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.cpp:
+ * tests/examples/overlay/qtgv-xoverlay.h:
+ examples: add video overlay examples for gtk, qt and qt graphics view
+ Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
+ adds all boilerplate to configure for using c++. The qt based examples are
+ optional like their gtk counterparts.
+
+2010-02-16 17:20:01 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-15 11:11:04 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/gio/gstgiosink.c:
- * ext/gio/gstgiosrc.c:
- Make the documentation of giosink/giosrc complete, large parts
- are based on the gnomevfssink/gnomevfssrc docs.
+ * 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.
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-16 10:09:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/gst-plugins-base-plugins.args:
- * 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.
+ * 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).
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-08 13:54:57 +0200 Kaj-Michael Lang <milang@tal.org>
- * 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)
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Handle stm module format
+ Fixes #609314.
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 12:10:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@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).
+ * ext/vorbis/gstivorbisdec.c:
+ ivorbisdec: set rank to SECONDARY
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 12:09:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@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.
+ * configure.ac:
+ * 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.
-2008-02-11 Julien Moutte <julien@fluendo.com>
-
- Patch by: Alan Peevers <peeves@pacbell.net>
+2010-02-03 14:37:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
- lock when calling alsa methods.
+ * 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.
-2008-02-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-15 10:23:13 +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).
+ * tests/check/elements/playbin.c:
+ playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-15 09:04:17 +0100 Sebastian Dröge <sebastian.droege@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/gstoggparse.c:
+ oggparse: Fix another format string compiler warning
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-15 08:56:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Branko Čibej <brane at xbc dot nu>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix format string compiler warnings
- * sys/xvimage/xvimagesink.c:
- Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
- Fixes bug #515654.
+2010-02-15 08:48:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-09 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Post a missing element message and an error message if no uridecodebin can be found
- * 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.
+2010-02-15 08:46:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * 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.
+2010-02-15 08:28:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-02-08 Wim Taymans <wim.taymans@collabora.co.uk>
+ * tests/check/elements/playbin2.c:
+ playbin2: Enable all unit tests
+ They're all working and valgrind clean now.
- * 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>
+2010-02-15 08:26:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Revert last change as we now check in gtk-doc.m4 for sed.
+ * 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.
-2008-02-08 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-02-15 08:20:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Find and subst SED when building the docs.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
+ and not CORE MISSING_PLUGIN.
-2008-02-08 Julien Moutte <julien@fluendo.com>
+2010-02-15 08:18:06 +0100 Sebastian Dröge <sebastian.droege@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/playback/gstplaybin2.c:
+ playbin2: Free the subtitle URI
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 08:06:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Make DISABLE_DEPRECATED defined *only* during CVS, not during
- pre-releases or releases.
+ * 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.
-2008-02-08 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 06:50:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * ext/gio/Makefile.am:
- Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
- reporting
+ * tests/check/elements/.gitignore:
+ tests: Add decodebin2 test to .gitignore
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 01:18:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/plugins/Makefile.am:
- Add the headers which need scanning for the GIO plugin. The rest of
- the docs still need migrating.
+ * 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.
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 01:17:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/Makefile.am:
* tests/check/Makefile.am:
- * tests/check/pipelines/.cvsignore:
- Add gio in a few more places.
+ * 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.
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-15 00:11:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * configure.ac:
- * ext/Makefile.am:
- * tests/check/Makefile.am:
- Move gio plugin from -bad and mark as experimental.
+ * ext/ogg/gstoggstream.c:
+ ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
+ See #609252.
-2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-14 23:16:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 96dc793 to 44ecce7
-2008-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-14 23:10:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix pbutils header.
+ * 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.
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-14 22:16:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Fix compiler warning.
+ * tests/check/elements/playbin.c:
+ playbin: Fix indention in the unit test
-2008-02-07 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-13 01:08:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/volume/gstvolume.c:
+ volume: Replace this variables by self
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
- Clear the addrinfo struct using memset. Fixes #514937.
+2010-02-12 19:43:13 +0100 Josep Torra Valles <n770galaxy@gmail.com>
-2008-02-06 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * gst/tcp/gstfdset.h:
- Remove unused field to same some memory.
+2010-02-12 18:00:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
- Mark action signals as such.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: fix Since tag
-2008-02-06 Michael Smith <msmith@fluendo.com>
+2010-02-12 14:19:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoradec.c: (_theora_granule_frame),
- (_inc_granulepos):
- Increment granulepos for new-bitstream versions appropriately.
- Fixes #514623.
+ * gst-libs/gst/riff/riff-read.c:
+ riff: treat JUNQ chunks like JUNK chunks
-2008-02-04 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-12 14:29:18 +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.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update basesrc segment duration and post duration messages from the streaming thread
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-11 14:10:02 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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-libs/gst/tag/tags.c:
+ tags: improve docs about determining the encoding
-2008-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2010-02-11 14:09:05 +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-libs/gst/tag/gstvorbistag.c:
+ comment: fix wrong header comment
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-01 13:50:14 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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-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.
-2008-02-02 Sebastian Dröge <slomo@circular-chaos.org>
+2010-02-01 19:01:33 +0100 Wim Taymans <wim.taymans@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.
+ * 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
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-01 18:56:34 +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.
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: cleanups
+ Avoid some typechecks.
+ Avoid dereferencing appsrc->priv all the time.
- * 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>
+2010-02-01 18:55:39 +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:
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: cleanups
+ Avoid some typecasting.
+ Avoid dereferencing appsink->priv all the time.
+
+2010-02-01 15:09:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: avoid some typecasts
+
+2010-01-29 16:34:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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
+
+2010-02-10 16:05:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2010-02-10 10:12:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back
+
+2010-02-09 17:39:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Update segment duration and post a duration message if the duration changes
+ Fixes bug #609423.
+
+2010-02-11 10:56:17 +0100 Benjamin Otte <otte@redhat.com>
+
+ * tests/examples/seek/Makefile.am:
+ build: link to libm in examples that use it
+ This fixes build failure in Fedora 13.
+
+2010-02-11 01:11:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * MAINTAINERS:
+ Update MAINTAINERS, add myself
+
+2010-02-11 23:57:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: back to development
+ Slushy freeze remains in effect.
+
+=== release 0.10.26 ===
+
+2010-02-10 20:17:36 +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:
- First round of plugin docs cleansups.
-
* 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-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:
- 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-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.26
-2008-02-01 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-02-10 20:16:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- * 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()
+2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.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.
- * configure.ac:
- Back to CVS
+2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-=== release 0.10.17 ===
+ * ext/ogg/gstoggstream.c:
+ oggdemux: use the default granpos functions for kate streams
+ Set timestamps on kate packets. See bug #600929.
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- releasing 0.10.17, "Peanut Butter and Jelly"
-
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.25.3 pre-release
- * gst-libs/gst/interfaces/mixeroptions.c:
- * gst-libs/gst/interfaces/mixertrack.c:
- Also remove the conditional registration of the signals
- that disappeared with the ABI change in 0.10.14
+2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * po/bg.po:
+ po: update translations
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- Revert patch to gstrtspconnection.c for brown paper bag
- release of -base. Re-opens: #511825
+2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-30 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * gst-libs/gst/interfaces/mixeroptions.h:
- * gst-libs/gst/interfaces/mixertrack.h:
- Change the way these deprecated function pointers are removed
- so that the compiled ABI is unconditionally smaller. This
- sets in stone an ABI break that actually occurred when the
- things were deprecated in 0.10.14, which seems to be the best
- fix as the only known users are oss-mixer and sunaudio-mixer in
- gst-plugins-good.
- Fixes: #513018
+2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2008-01-30 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * win32/common/libgstpbutils.def:
- Export the two new _get_type() functions which are needed
- by the python bindings.
+2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
- Cast glong to time_t as time_t might have a different type on
- other platforms, like FreeBSD, and we get a compiler warning
- otherwise. Fixes bug #511825.
+2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-29 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init):
- Initialize the GstRingerBuffer class to get it's debug category
- initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
- category and otherwise we get some g_critical(). Fixes bug #512334.
+2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+ * common:
+ Automatic update of common submodule
+ From 15d47a6 to 96dc793
- * configure.ac:
- Back to CVS
+2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.10.16 ===
+ * 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
-2008-01-28 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- releasing 0.10.16, "Scheduled Interruption"
+ 0.10.25.2 pre-release
-2008-01-22 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- 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.
+ * 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
-2008-01-18 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@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:
- Oops - add new sk.po mentioned in the LINGUAS I just committed
+ * 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
-2008-01-17 Jan Schmidt <jan.schmidt@sun.com>
+2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * po/LINGUAS:
- Add ca translation to the disted list.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
- * win32/vs6/libgstsdp.dsp:
- Convert line endings to CRLF
+2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-17 Sebastien Moutte <sebastien@moutte.net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't skip an element when getting the topology
+ Fixes #608167
- * win32/MANIFEST:
- Add win32/vs6/libgstrtsp.dsp to MANIFEST
+2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+ * 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.
- * win32/common/libgstsdp.def:
- * win32/common/libgstvideo.def:
- Add new API declarations
+2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: properly set up the media type for kate streams
+ See #600929.
- * 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.
+2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
-2008-01-14 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.
- * 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.
+2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-13 David Schleef <ds@schleef.org>
+ * 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
- * gst/videoscale/vs_4tap.c:
- Fix valgrind error on 4tap scaling method.
+2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-13 Sebastien Moutte <sebastien@moutte.net>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: don't push headers twice
+ Don't push the stream headers twice but only in the activation of a chain.
+ Fixes #607929
- * 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.
+2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-13 Julien Moutte <julien@fluendo.com>
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
- * 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.
+2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+ * 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.
- * tests/check/Makefile.am:
- Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
+2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
- * 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.
+ * 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.
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
+2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/Makefile.am:
- Fix include path order
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinder for ISO MP4 files
+ Fixes bug #607848.
-2008-01-11 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-01-24 13:29:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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
-2008-01-11 Michael Smith <msmith@fluendo.com>
+2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Strip trailing \0 for subtitle OGM streams
+ Fixes bug #607870.
-2008-01-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@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):
- 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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Correctly set DELTA_UNIT flag for OGM streams
-2008-01-10 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Extract tags from OGM text streams and don't push them downstream
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description strings for ID3, APE, and ICY tags.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
-2008-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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>
+ * 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.
- Patch by: Bastien Nocera <hadess at hadess net>
+2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
- (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
- Use snd_mixer_selem_set_{playback|capture}_volume_all() if
- the volume is the same for all channels. This works around
- some problem in alsa that leaves us with inconsistent state
- for some reason (#486840).
+ * gst/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.
-2008-01-07 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
- Patch by: Jerone Young <jerone at gmail com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for Zip Block Motion Video
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
- If there's no mixer track by the name of 'Master' or 'Front',
- check if there's one called 'PCM' before trying the generic
- fallback logic (fixes #506928, where we pick 'Mic' as master
- track for the AD1984 card in a Thinkpad T61/X61 laptop).
+2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-07 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Zip Block Motion Video
- * gst/playback/gstplay-enum.c:
- (register_gst_autoplug_select_result),
- (gst_autoplug_select_result_get_type), (register_gst_play_flags),
- (gst_play_flags_get_type):
- * gst/playback/gstplay-enum.h:
- Add enums for configuration flags.
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (no_more_pads_cb),
- (autoplug_select_cb), (gst_play_bin_change_state):
- Merge mode with flags.
- Add more property getters/setters, defaults and docs.
- Add properties to get number of audio/video/text streams.
- Create sink object in _init so that we can always rely on it being
- there.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_init),
- (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
- (activate_vis), (gst_play_sink_reconfigure),
- (gst_play_sink_set_flags), (gst_play_sink_get_flags),
- (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Use flags to configure the sink pipelines.
- Add tee before audio pipeline so that we can use it for visualisations.
- Start working on integrating visualisations.
- Remove mode, we can do everything with the flags now.
- Add method to configue the sink pipeline.
+2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-06 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: YUNV is a fourcc which is also used for YUY2 raw video
- * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
- (check_buffer_timestamp), (check_buffer_duration):
- Turn these functions into macros so we can see right away
- where the failure occured.
+2010-01-23 15:13:45 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-05 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: vp61 and VP61 are also valid On2 VP6 fourcc
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
- debugging information to understand how X calculates the stride
- for XvImages.
+2010-01-23 15:10:45 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for On2 VP5
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (volume_choose_func),
- (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
- (volume_setup):
- * gst/volume/gstvolume.h:
- Use GstAudioFilter as base class for the volume element instead of
- plain GstBaseTransform.
+2010-01-23 15:04:35 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-03 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
- Don't set element details for the abstract GstAudioFilter class.
+2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-02 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for LOCO Lossless codec
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
- Implement get_unit_size() vmethod of GstBaseTransform.
+2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
-2008-01-01 Edward Hervey <edward.hervey@collabora.co.uk>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for LOCO Lossless codec
- * gst-libs/gst/pbutils/Makefile.am:
- * gst-libs/gst/pbutils/pbutils.h:
- Use glib-enum generator to have a proper enum GType for
- GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
+2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
- * tests/check/Makefile.am:
- * tests/check/pipelines/theoraenc.c:
- Reenable theoraenc test, which fails on the buildbot but
- not locally.
+2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Autodesk Animator codec
- * docs/libs/.cvsignore:
- * docs/plugins/.cvsignore:
- Add *-undeclared.txt to fix buildbot.
+2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-12-31 David Schleef <ds@schleef.org>
+ * gst-libs/gst/riff/riff-media.c:
+ riff: Add mapping for Autodesk Animator Codec
- * tests/check/Makefile.am:
- Second attempt at disabling theoraenc test long enough to
- get buildbot to compile -base.
+2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: ...and set caps on queued packet buffers too
- * tests/check/pipelines/theoraenc.c:
- Disable theoraenc test long enough to get the buildbot to
- compile a recent -base.
+2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-12-31 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Set caps on header buffers
- * tests/examples/seek/seek.c: (stop_cb):
- Make sure we reset the slider value to 0.0 without racing against a
- possible g_idle that sets it to something else.
+2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-31 Thijs Vermeir <thijsvermeir@gmail.com>
+ * 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
- * sys/ximage/ximagesink.c:
- fix typo
+2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-12-30 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: include stdio.h for sscanf
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- Add Location header so that we can start implementing redirects.
- See #506025.
+2010-01-22 01:49:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: add PNM typefinder
+ Add PNM typefinder, so we can remove the one that's in the PNM plugin
+ in -bad (which btw uses different/wrong media types that don't match
+ the ones used by gdkpixbufdec) and people don't make fun of us for
+ loading image decoders when typefinding and playing back audio files.
- * gst/subparse/gstssaparse.c:
- combine if's
+2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
-2007-12-29 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: rename performance category
+ rename the performance category to ffmpegcolorspace_performance
+ as there is already a global GST_CAT_PERFORMANCE in core
- * gst/subparse/gstssaparse.c:
- remove duplicate log message
-
-2007-12-29 Sebastian Dröge <slomo@circular-chaos.org>
-
- * ext/libvisual/visual.c: (gst_visual_chain):
- Fix 'xyz may be used uninitialized' compiler warnings caused
- by broken g_assert_not_reached() macro in GLib-2.15.x and don't
- abort() in any case but properly report the error.
-
-2007-12-28 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
- (gst_play_bin_finalize), (gst_play_bin_set_uri),
- (gst_play_bin_set_suburi), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
- (autoplug_select_cb), (activate_group), (deactivate_group),
- (setup_next_source), (save_current_group),
- (gst_play_bin_change_state):
- Code cleanups.
- Remove next-uri, we can use the uri property just fine.
- Fix some crasher.
- Unref uridecodebin when switching.
- Fix going to READY.
-
- * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
- (gst_play_sink_init), (gst_play_sink_dispose),
- (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
- (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
- (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
- (gst_play_sink_set_property), (gst_play_sink_get_property),
- (gen_video_chain), (gen_text_element), (gen_audio_chain),
- (gen_vis_element), (gst_play_sink_get_mode),
- (gst_play_sink_set_mode), (gst_play_sink_set_flags),
- (gst_play_sink_get_flags), (gst_play_sink_request_pad),
- (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
- (gst_play_sink_change_state):
- * gst/playback/gstplaysink.h:
- Add some locking to make things threadsafe.
+2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/test7.c: (about_to_finish_cb):
- Fix test.
+ * 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.
-2007-12-22 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
- (gst_video_scale_get_property), (gst_video_scale_transform_caps),
- (gst_video_scale_transform):
- Don't claim to be able to handle/transform caps that can't really
- be handled by the currently selected scaling method (here: RGB or
- packed YUV with 4-tap method). Also add locking to method property.
+ * 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.
- * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
- (test_basetransform_based):
- Some test pipelines for the above (not entirely valgrind clean yet
- apparently).
+2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-21 David Schleef <ds@schleef.org>
+ * ext/ogg/gstoggstream.c:
+ oggstream: avoid division by 0 on bad annodex streams
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Add additional RGBA and RGB-24 video formats.
+2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for y4m container
- * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
- (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
- (test_suburi_error_wrongproto), (test_missing_primary_decoder):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
- (cddabasesrc_suite):
- Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
- deprecated in the future (see #498924).
+2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: ptime/maxptime should be unsigned
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
- * gst/playback/gststreamselector.c: (gst_selector_pad_event):
- Don't leak event.
+2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-12-20 Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: ptime should be in nanoseconds
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
- * gst-libs/gst/riff/riff-read.c:
- Use GST_ROUND_UP_2 macro
+2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 14cec89 to 15d47a6
- * gst/playback/.cvsignore:
- Ignore more.
+2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
-2007-12-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: rewrite h.264 detection
+ Make detection simpler: check for NALs, check that they make
+ sense, and report how certain we are that it's a raw H.264 stream.
+ Fixes: #583376.
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
- * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
- (set_active_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (setup_sinks), (playbin_set_subtitles_visible):
- Make switching off of subtitles work. To avoid all kind of
- problems with unlinking of the subtitle input, we just keep
- the subtitle inputs linked as they are and tell textoverlay
- not to render them. Fixes #373011.
- Other subtitle switching issues (esp. when there are both
- external and in-stream subtitles) remain. They'll be solved
- in playbin2.
+2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Reject empty caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=607353
- * gst/playback/gststreamselector.c: (gst_selector_pad_init):
- Init the pad segment too.
+2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-12-18 Wim Taymans <wim.taymans@collabora.co.uk>
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: No need to subtract begin time
+ Last stop is already based on the chain start and there is no need
+ to subtract the chain start as it may lead to a negative overflow.
+ This was causing seeking issues when the target chain was not
+ the first one (that has chain start = 0)
+ Fixes #606382
- * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
- (gst_audioringbuffer_open_device),
- (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release), (gst_audioringbuffer_start),
- (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
- (gst_audio_sink_create_ringbuffer):
- Improve debug output.
+2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (gst_ring_buffer_delay):
- Prevent some functions from doing things and failing when the
- ringbuffer is not yet acquired.
+ * gst-libs/gst/audio/audio.h:
+ audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
+ Fixes bug #607381.
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/interfaces/interfaces.h:
- Also remove interfaces.h from CVS as it is not needed anymore.
+ * 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
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/interfaces/Makefile.am:
- interfaces.h is not used anymore so remove it from the build
- process.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for MXF container format
-2007-12-17 David Schleef <ds@schleef.org>
+2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- Add a "blink" pattern. Turn on the pain. Apologies. It's useful
- for testing vertical refresh synchronization.
+ * gst/playback/gstplaysink.c:
+ playsink: re-use iterator callback to avoid code duplication
-2007-12-17 David Schleef <ds@schleef.org>
+2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/video/video.c:
- * gst-libs/gst/video/video.h:
- Add new GstVideFormat enum and write a bunch of helper functions
- based around it.
+ * gst/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).
-2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * Makefile.am:
- Use new common/win32.mak.
+ * 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.
-2007-12-17 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Add debug info.
- When going from PLAYING to PAUSED, pause the ringbuffer before calling
- the parent state change function, just like the audiosink, because the
- parent waits for the element to finish its processing before completing
- the state change. This makes going to PAUSED a lot snappier.
- When going from READY to PAUSED, don't allow the ringbuffer to start
- yet.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: sprinkle some more locking
+ ... to avoid races and ensure some data structure consistency.
+ See also #574289.
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Yet another fix for broken software that produce files with an empty
- blockalign field. Instead of completely failing, make a second attempt
- at guessing the width/depth by looking at strf->size.
+ * 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.
-2007-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
- (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
- * gst-libs/gst/pbutils/install-plugins.c:
- (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_plugin_message_get_installer_detail),
- (gst_missing_encoder_installer_detail_new):
- * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
- * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
- Turn a few g_assert_not_reached() into g_return_val_if_reached() to
- avoid compiler warnings (#503930).
+ * 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.
-2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
- for jpeg video streams.
- Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
- for the above modification.
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix pad ref leak
-2007-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
- (gst_x_overlay_handle_events):
- More guards (we don't want klass to end up being NULL).
+ * docs/plugins/Makefile.am:
+ docs: fix out-of-source build
-2007-12-14 Sebastian Dröge <slomo@circular-chaos.org>
+2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- * gst/volume/gstvolume.c: (gst_volume_init):
- Use new gst_base_transform_set_gap_aware() function as volume
- correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
- for this.
+ * tests/icles/stress-playbin.c:
+ stress-playbin: fix error return check
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (msg_segment_done), (main):
- Don't go to READY on EOS as this avoids testing of seeking and
- restarting after EOS, use the stop button when you want to READY.
- Don't try to do a flushing seek in segment-done, it does not make
- sense to use this for gapless playback and is not needed.
+ * ext/theora/Makefile.am:
+ * ext/theora/gsttheora.c:
+ * ext/theora/gsttheoradec.c:
+ * ext/theora/gsttheoraenc.c:
+ * ext/theora/gsttheoraparse.c:
+ * ext/theora/theora.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c:
+ theora: Rename source files to have the same name as the headers
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
- (reset_rate_timer), (update_in_rates), (update_out_rates),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_chain), (gst_queue_loop):
- Use separate timers for input and output rates.
- Pause measuring the output rate when we block for more data.
- See #503262.
+ * ext/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
-2007-12-14 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_chain):
- Pause the timer to measure the input rate when we block because the
- queue is filled. See #503262.
+ * ext/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.
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis com>
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: Add official 6.1 and 7.1 channel mappings
+ These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_free):
- Close control sockets. Fixes #503440.
+2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.com>
-2007-12-13 Wim Taymans <wim.taymans@collabora.co.uk>
+ * 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.
- * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
- Expose the right pad in the right place with the right element.
+2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: mp4 video is not parsed
- * gst-libs/gst/pbutils/descriptions.c: (formats):
- Add description for 'private' dts caps (who come up with that name?).
+2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Add aac stream-format to caps
+ Also add the aac stream-format field on the caps when
+ detecting it.
- * Makefile.am:
- Add check-exports target and run it with 'make check'.
+2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
- * 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/playback/gstplaysink.c:
+ playsink: Fix handling of the native audio/video flags
+ Fixes bug #606687.
- * gst-libs/gst/cdda/base64.h: (rfc822_binary):
- * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
- Change internal functions to __gst_foo so they dont' get exported.
+2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
- * win32/common/libgstaudio.def:
- Add missing symbols.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Fix unitialized variable.
+ If the package isn't handled, gracefully return GST_FLOW_OK.
-2007-12-11 David Schleef <ds@schleef.org>
+2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ChangeLog: remove conflict markers
+ * 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).
-2007-12-11 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/gnomevfs/Makefile.am:
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
- Use gst_tag_freeform_string_to_utf8() here, which also takes
- into account any character sets specified by the user via
- environment variables.
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ docs: minor netbuffer documentation fix
-2007-12-10 Wim Taymans <wim.taymans@gmail.com>
+2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/Makefile.am:
- Also link to libm.
+ * 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.
-2007-12-10 Wim Taymans <wim.taymans@gmail.com>
+2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- No need for floating point operations here. avoids having to link
- against the math library too.
+ * 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
-2007-12-10 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (formats),
- (format_info_get_desc):
- * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
- (GST_START_TEST):
- Add one or two missing formats. Generate ADPCM description
- dynamically depending on layout/format.
+ * gst-libs/gst/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.
-2007-12-09 Sebastian Dröge <slomo@circular-chaos.org>
+2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
- * configure.ac:
- Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-plugins-base.spec.in:
+ Add missing source file for tagger to Makefile and update spec file
-2007-12-08 Tim-Philipp Müller <tim at centricular dot net>
+2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
- Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
+ * gst-libs/gst/riff/riff-media.c:
+ riff-media: handle 32 bit raw RGB video.
- * 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).
+2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/elements/subparse.c: (srt_input), (srt_input0),
- (test_src):
- Add unit test for the above.
+ * ext/ogg/gstoggstream.c:
+ oggdemux: decide flac header packet by content rather than count
-2007-12-06 Wim Taymans <wim.taymans@gmail.com>
+2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * gst/playback/gstplay-enum.c:
- (register_gst_autoplug_select_result),
- (gst_autoplug_select_result_get_type):
- * gst/playback/gstplay-enum.h:
- Add missing files.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: reset header packet count at bos page
-2007-12-05 Wim Taymans <wim.taymans@gmail.com>
+2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: add support for buffer-lists
- * 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.
+2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Ignore zero framerate
+ https://bugzilla.gnome.org/show_bug.cgi?id=606163
-2007-12-03 Edward Hervey <bilboed@bilboed.com>
+2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
+ * 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
-2007-12-03 Wim Taymans <wim.taymans@gmail.com>
+2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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
+ * 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
-2007-11-30 Wim Taymans <wim.taymans@gmail.com>
+2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: Accept maxptime from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
- * 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.
+2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-11-26 Stefan Kost <ensonic@users.sf.net>
+ * ext/ogg/gstoggstream.c:
+ oggdemux: enhance flac packet duration calculation
- * gst/audiotestsrc/gstaudiotestsrc.c:
- * gst/volume/gstvolume.c:
- * gst/volume/gstvolume.h:
- Add GAP-flag support.
+2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-11-24 Julien MOUTTE <julien@moutte.net>
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
- * 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.
+2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * 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.
-2007-11-23 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (audioresample_query), (audioresample_query_type),
- (gst_audioresample_set_property):
- Implement latency query.
+ * 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.
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain):
- Our EOS time contains the base_time, _wait_eos() expects a running_time
- so we have to subtract the base_time again before calling the function.
- This fixes an EOS regression where the base_time was added twice and EOS
- took longer and longer in certain situations.
- Fixes #498767.
+ * 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.
-2007-11-21 Wim Taymans <wim.taymans@gmail.com>
+2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@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()
+ * tests/icles/stress-xoverlay.c:
+ tests: fix warning and memory leak in stress-overlay test
+ Not all messages have structures and we need to unref messages
+ when returning GST_BUS_DROP in the sync bus handler.
- * gst-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()
+2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-11-20 Wim Taymans <wim.taymans@gmail.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: correctly eat empty and dummy buffers
- Patch by: Joe Peterson <lavajoe at gentoo dot org>
+2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix compilation on FreeBSD (Gentoo). Fixes #498228.
+ * 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.
-2007-11-19 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/libs/fft.c: (GST_START_TEST):
- Fix scaling to really have dB instead of something else.
+ * 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
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): There's a nice macro to check
- GTK version, use it.
+ * gst/adder/gstadder.c:
+ adder: fix juvenile comment
-2007-11-19 Julien MOUTTE <julien@moutte.net>
+2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (main): Try to support stable version
- of GTK.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix typo in debug message
-2007-11-17 Stefan Kost <ensonic@users.sf.net>
+2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/README:
- * gst/playback/test7.c:
- Fix the build + little README update.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some type checks
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
- Add playbin2 seek pipeline.
+ * gst/playback/gstplaybin2.c:
+ playbin2: avoid leaking selector request pads
-2007-11-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstplayback.c: (plugin_init):
- * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
- (eos_cb), (about_to_finish_cb), (main):
- Add playbin2.
- Added gapless playback example.
+ * gst/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.
- * 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).
+2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: don't name the queue
+ There is no reason to name the queue.
+ Fixes #605219
- * gst-plugins-base.doap:
- Oops, fix the release name.
+2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * win32/common/libgstrtp.def:
+ defs: update defs with new symbols
- * gst-plugins-base.doap:
- Add 0.10.15 release
+2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-11-16 Jan Schmidt <jan.schmidt@sun.com>
+ * 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
- * configure.ac:
- Back to CVS
+2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-=== release 0.10.15 ===
+ * common:
+ Automatic update of common submodule
+ From 47cb23a to 14cec89
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * configure.ac:
- releasing 0.10.15, "No need to argue"
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: add Since marker for the new tolerance property
-2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/vs6/libgstfft.dsp:
- Convert line endings to DOS.
+ * gst-libs/gst/tag/lang.c:
+ docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
-2007-11-15 Sebastien Moutte <sebastien@moutte.net>
+2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * 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.
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
- * win32/vs6/libgstinterfaces.dsp:
- * win32/vs6/libgstrtsp.dsp:
- Convert line endings back to DOS
+ * 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.
-2007-11-14 Jan Schmidt <jan.schmidt@sun.com>
+2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/fft/kiss_fft_f32.h:
- * gst-libs/gst/fft/kiss_fft_f64.h:
- * gst-libs/gst/fft/kiss_fft_s16.h:
- * gst-libs/gst/fft/kiss_fft_s32.h:
- Don't include malloc.h which doesn't exist on Mac OSX.
- Instead, pull in glib.h and use g_malloc/g_free for
- consistency. Fixes: #496548
+ * gst-libs/gst/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.
-2007-11-09 Stefan Kost <ensonic@users.sf.net>
+2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstdecodebin2.c:
- Dont leak ghostpad. Fixes #475451.
+ decodebin2: add some debugging
-2007-11-09 Wim Taymans <wim.taymans@gmail.com>
+2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * docs/design/design-decodebin.txt:
- * gst/playback/gstdecodebin2.c: (analyze_new_pad):
- Update some more docs and comments.
+ * 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
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Patch by: Sebastien Moutte <sebastien moutte net>
+ * 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/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).
+2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstaudio.dsp:
- * win32/vs6/libgstrtp.dsp:
- * win32/vs6/libgsttag.dsp:
- Update vs6 projects files (#494346).
+ * gst/audiorate/Makefile.am:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ audiorate: use separate header file
-2007-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: set DISCONT when resyncing (e.g. newsegment)
- * 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).
+2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-11-06 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: also fill up segments if possible
- * 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.
+2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-11-06 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/playback/gstplaybasebin.c: (queue_threshold_reached),
- (finish_source):
- Avoid crash when there are external subtitles (fixes #491722).
+2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-11-03 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: properly report truncated samples as dropped samples
- * 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.
+2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/tag/lang.c:
+ docs: mention that gst_tag_get_language_name() may return NULL
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * win32/common/libgstrtp.def:
- Add some more missing symbols (#492813).
+ * tests/check/libs/tag.c:
+ checks: some more testing for the new language code functions
-2007-11-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/mixeroptions.c:
+ * gst-libs/gst/interfaces/mixertrack.c:
+ docs: misc. mixer docs improvements
- * 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.
+2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-11-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ docs: add short descriptions for API reference contents page
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
- * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
- Include our own _stdint.h instead of sys/types.h, makes MingW happy
- (#492306).
+ * gst-libs/gst/tag/lang-tables.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ tag: make internal language names table static
- * 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).
+2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/sdp/Makefile.am:
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix includes and LIBS for win32/Mingw (#492306).
+ * gst-libs/gst/tag/lang.c:
+ * gst-libs/gst/tag/mklangtables.c:
+ tag: don't use GLib 2.22 API
+ g_mapped_file_unref() was introduced in GLib 2.22, but we depend
+ only on GLib 2.18, so use g_mapped_file_free() when compiling
+ against older GLib versions until we bump the GLib dependency.
- * tests/examples/dynamic/addstream.c (pause_play_stream):
- Use more portable g_usleep() instead of sleep() (#492306).
+2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-11-01 Tim-Philipp Müller <tim at centricular dot net>
+ * .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()
- Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * ext/ogg/gstoggstream.c:
+ ogg: ogm video has constant packet duration
- * gst-libs/gst/audio/gstringbuffer.h:
- No trailing commas in enum list (for gcc-2.9x).
+2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
- * 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).
+ * ext/ogg/gstoggstream.c:
+ oggdemux: implement old fLaC mapping
- * win32/common/config.h.in:
- Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
+2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * win32/common/libgstinterfaces.def:
- * win32/common/libgstrtp.def:
- Export a few more symbols (#492114).
+ * 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
-2007-11-01 Stefan Kost <ensonic@users.sf.net>
+2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- Readd the deprecation guards, but preserve compilability.
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: remove redundant fields
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
- (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
- Preserve channel layout when fixating the number of channels in the
- output caps, or make sure there's a suitable channel position layout
- set on the caps if required. Fixes #430677.
+ * 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.
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/vorbis/vorbisdec.c:
+ vorbisdec: fix peer query
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/theora/theoradec.c:
+ theoradec: fix query
-2007-10-31 Stefan Kost <ensonic@users.sf.net>
+2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * ext/theora/theoradec.c:
+ theoradec: small cleanups
-2007-10-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/libs/audio.c: (init_value_to_channel_layout),
- (test_channel_layout_value_intersect), (audio_suite):
- Add simple unit test to make sure GstValue intersection
- of channel layouts works the way I think it does.
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: use gst_pad_peer_query()
-2007-10-30 Stefan Kost <ensonic@users.sf.net>
+2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudiofilter.h:
- * gst-libs/gst/interfaces/mixer.h:
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/sdp/gstsdpmessage.c:
- Fix the docs according to what gtk-doc complained about.
+ * gst/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.
-2007-10-30 Stefan Kost <ensonic@users.sf.net>
+2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/stress-playbin.c:
- Fix the build.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only flush downstream on seeks for flushing seeks
-2007-10-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-30 Wim Taymans <wim.taymans@gmail.com>
+2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-28 Sebastian Dröge <slomo@circular-chaos.org>
+2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-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).
+2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
- Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: improve keyframe seeking
+ Improve keyframe seeking.
+ Fix reverse playback.
- * 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).
+2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-10-24 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.
- Inspired by patch of: René Stadler <mail at renestadler dot de>
+2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaysink.c:
+ playsink: Some minor cleanup
-2007-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't abort with an assertion if we receive a seek event with
- a start type of NONE (see launchpad bug #155878).
+ * gst/playback/gstplaybin2.c:
+ playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
-2007-10-22 Wim Taymans <wim.taymans@gmail.com>
+2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add comment
- * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
- Remove unneeded lock causing a race in typefind and shutting down.
- Fixes #485753.
+2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-16 Wim Taymans <wim.taymans@gmail.com>
+2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-10-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/stress-playbin.c:
- Does what it says on the tin.
+ * 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.
-2007-10-15 Wim Taymans <wim.taymans@gmail.com>
+2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
- * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
- Fix queue negotiation. See #486758.
+ * 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.
-2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
+2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
- (gst_xvimagesink_xwindow_new),
- (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
+ * ext/ogg/gstoggstream.c:
+ oggdemux: fix for I-frame-only theora
- 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.
+2009-12-03 01:16:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ ogg: log when ogg mapper doesn't accept the setup header packet
- * gst-libs/gst/tag/gstvorbistag.c:
- * tests/check/libs/tag.c:
- Extract vorbis comment LICENSE tags correctly.
+2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-10-11 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggstream.c:
+ ogg: extract width, height and PAR from theora header and add to caps
- Patch by: Jason Kivlighn <jkivlighn gmail com>
+2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c:
- * tests/check/libs/tag.c:
- Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
+ * ext/ogg/gstoggstream.c:
+ ogg: extract number of channels from FLAC, speex and vorbis headers
+ Because we can.
-2007-10-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't error out when a buggy downstream element doesn't
- handle the newsegment event we send properly (especially
- not without posting a meaningful error message on the
- bus). See bug #471370 and launchpad bug #136264.
+ * gst/playback/gstplaybin2.c:
+ build: fix build with debug logging disabled.
-2007-10-10 Wim Taymans <wim.taymans@gmail.com>
+2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_drain):
- Use new basesink method to make our EOS drain interruptable.
+ * 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’
-2007-10-10 Jan Schmidt <Jan.Schmidt@sun.com>
+2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Fix silly search-replace oversight.
+ * 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.
-2007-10-09 Wim Taymans <wim.taymans@gmail.com>
+2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+ * 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-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
- (gst_basertppayload_set_outcaps):
- Fix caps memleak. Fixes #484989.
+2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
+ * gst/videoscale/vs_4tap.c:
+ Build fix for MSVC
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
- Fix debug output.
+ * gst/subparse/qttextparse.c:
+ build: add missing includes for sprintf and atoi
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * 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.
+ * 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
-2007-10-08 Jan Schmidt <Jan.Schmidt@sun.com>
+2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtppayloads.c:
- Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
- to avoid compiler warnings
+ * 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>
-2007-10-08 Wim Taymans <wim.taymans@gmail.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: conditionally cleanup sami context
+ Only cleanup sami context if we are parsing sami subtitles,
+ otherwise we might have crashes.
- * 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.
+2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-10-08 Wim Taymans <wim.taymans@gmail.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
- * 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.
+2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From 87bf428 to 47cb23a
- * gst-libs/gst/tag/gsttagdemux.c:
- Don't leak caps.
+2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-10-06 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Automatic update of common submodule
+ From da4c75c to 87bf428
- * 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.
+2009-11-30 10:22:15 +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.
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Fix some pad refcount issues
+ Fixes bug #603345.
-2007-10-05 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- (gst_rtp_buffer_get_payload_subbuffer):
- Fix bug introduced with last commit which inverted the logic and
- caused all buffers to be dropped. Fixes #483620.
- Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
+ * common:
+ Automatic update of common submodule
+ From 53a2485 to da4c75c
-2007-10-04 Stefan Kost <ensonic@users.sf.net>
+2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Replace g_return_if_val (as it could be disabled), with regular return
- and warning.
+ * ext/ogg/gstoggstream.c:
+ * ext/ogg/gstoggstream.h:
+ oggdemux: handle theora streams with 0 keyoffset
-2007-10-03 Stefan Kost <ensonic@users.sf.net>
+2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
- * tests/check/pipelines/simple-launch-lines.c:
- Print message name and not just number.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle unknown streams
-2007-10-02 Wim Taymans <wim.taymans@gmail.com>
+2009-11-26 14:30:33 +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.
+ * 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.
-2007-10-02 Stefan Kost <ensonic@users.sf.net>
+2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/snapshot/snapshot.c:
- Fix the build.
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
+ ...and fix code style a bit.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Add RFC 3556 bandwidth modifiers.
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ audiofilter: Add _CAST variants of the cast macros
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-25 10:26:16 -0600 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/rtp/gstrtppayloads.c:
- Update documentation.
+ * 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.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
+ Fixes bug #602834 and #350748.
- * 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.
+2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
- (gst_sdp_parse_line):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Add some more comments.
+ * 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.
-2007-10-01 Wim Taymans <wim.taymans@gmail.com>
+2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
- * 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.
+ * 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.
-2007-09-29 Sebastien Moutte <sebastien@moutte.net>
+2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
- * 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>
+ * 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>
- * tests/examples/snapshot/snapshot.c: (main):
- Print error when pipeline failed to construct.
+ * 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.
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
- * 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.
+ * 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.
-2007-09-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
- * 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>
+ * 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>
- * configure.ac:
- Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
- of duplicating these macros in configure.ac.
+ * ext/ogg/gstoggmux.c:
+ oggmux: don't overwrite object properties
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
- * po/hu.po:
- * po/sv.po:
- * po/uk.po:
- Updated translations to 0.10.14
+ * 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.
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
- translated by: Jakub Bogusz <qboosh@pld-linux.org>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ debug: fix format string that was missing a var
- * po/pl.po:
- Added Polish translation.
+2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/adder/gstadder.c:
+ * tests/check/elements/adder.c:
+ adder: make events succeed, if they succed on atleast one pad
- translated by: Ilkka Tuohela <hile@iki.fi>
+2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
- * po/fi.po:
- Added Finnish translation.
+ * 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
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- translated by: Jorge González González <aloriel@gmail.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Fix stupid bug introduced in last commit
- * po/es.po:
- Added Spanish translation.
+2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- translated by: Mogens Jaeger <mogens@jaeger.tf>
+2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * po/da.po:
- Added Danish translation.
+ * 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.
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- translated by: Funda Wang <fundawang@linux.net.cn>
+ * common:
+ Automatic update of common submodule
+ From 0702fe1 to 53a2485
- * po/zh_CN.po:
- Added Chinese (simplified) translation.
+2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
-2007-09-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- translated by: Alexander Shopov <ash@contact.bg>
+2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * po/bg.po:
- Added Bulgarian translation.
+ * 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.
-2007-09-21 Wim Taymans <wim.taymans@gmail.com>
+2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_push_one):
- Fix compilation wrt printf arguments.
+ * 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.
-2007-09-20 Wim Taymans <wim.taymans@gmail.com>
+2009-11-15 19:36:21 +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.
+ * 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.
-2007-09-20 Stefan Kost <ensonic@users.sf.net>
+2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
- * tests/check/generic/states.c:
- Improved state change unit test.
+ * 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
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * docs/plugins/.cvsignore:
- * tests/check/.cvsignore:
- Ignore registries in any format.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix 'uninitialized' compiler warning
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@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.
+ * configure.ac:
+ configure: bump core requirement to 0.10.25.1
+ We depend on new API that's only in git so far.
-2007-09-19 Wim Taymans <wim.taymans@gmail.com>
+2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.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-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
-2007-09-19 Stefan Kost <ensonic@users.sf.net>
+2009-11-17 16:39:09 +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/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.
-2007-09-17 Jan Schmidt <Jan.Schmidt@sun.com>
+2009-11-17 16:37:57 +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/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/audiorate/gstaudiorate.c:
- Always pull in config.h before including any system headers.
+2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-09-17 Wim Taymans <wim.taymans@gmail.com>
+ * gst/audioconvert/gstaudioconvert.c:
+ audioconvert: remove unused array
- * 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.
+2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: Use new double->fraction transformation function from core
- * 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>
+2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/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.
-2007-09-15 Wim Taymans <wim.taymans@gmail.com>
+2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
- Add some debug info when negotiating caps.
+ * gst/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- A buffer with an empty payload is also a valid buffer.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't leak the GError and debug string when parsing error messages
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
- * 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.
+ * 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
-2007-09-14 Wim Taymans <wim.taymans@gmail.com>
+2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-09-14 Julien MOUTTE <julien@moutte.net>
+2009-11-11 17:08:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
- typefind for QCP files (RFC #3625)
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Only recreate factory caps if necessary and cache them
-2007-09-13 Wim Taymans <wim.taymans@gmail.com>
+2009-11-10 18:27:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/playback/gstsubtitleoverlay.c:
+ subtitleoverlay: Use gst_pad_get_caps_reffed()
-2007-09-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2007-09-11 Wim Taymans <wim.taymans@gmail.com>
+2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot 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/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.
+2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-11 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/playback/gstsubtitleoverlay.c:
+ * gst/playback/gstsubtitleoverlay.h:
+ subtitleoverlay: Set the video framerate on parsers if possible
+ Fixes bug #599649.
-2007-09-10 Wim Taymans <wim.taymans@gmail.com>
+2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Make fps a GstFraction typed property and use it properly
-2007-09-10 Stefan Kost <ensonic@users.sf.net>
+2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
- * gst/playback/gstdecodebin2.c:
- Don't leak request pads. Fixes #475395.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ subparse: Add property for the video framerate
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler dot de>
+ * 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.
- * 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.
+2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-09 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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/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.
+2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+ * 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-libs/gst/rtp/gstrtpbuffer.c:
- * tests/check/libs/rtp.c:
- Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
+2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
+2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * gst/playback/gstplaysink.c:
+ playsink: Add a queue before subtitleoverlay
+ This will improve playback, and the same thing is done
+ for subpicture streams too.
-2007-09-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/playback/gstplaysink.c:
+ playsink: Use subtitleoverlay for subtitles
-2007-09-06 Stefan Kost <ensonic@users.sf.net>
+2009-11-02 07:43:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Fix configure check for HAVE_LIBXML_HTML.
+ * 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
-2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/.cvsignore:
- Ignore more, in case the build bots work again one day.
+ * 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.
-2007-09-06 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
- Reviewed by: Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoradec.c:
+ theoradec: Keep timestamp from incoming buffer if it is valid
+ Fixes bug #601627.
- * 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.
+2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Integrate libgstfft into the docs.
+ * 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.
- * 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>
+2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- Need to include stdlib.h for abs() here too.
+ * 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.
-2007-09-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gststreaminfo.c:
- Fix 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.
-2007-09-05 Stefan Kost <ensonic@users.sf.net>
+2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gststreaminfo.c:
- Clean up some half-disabled code and comment.
+ * 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
-2007-09-04 Wim Taymans <wim.taymans@gmail.com>
+2009-11-10 00:48:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+ * gst/playback/gstplaysink.c:
+ playsink: assign chain->mute before using it
+ Fixes GObject warnings when starting totem.
- * 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.
+2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * 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/gstbasertppayload.c:
- Bump the MTU to 1400.
+2009-11-09 19:58:20 +0100 Edward Hervey <bilboed@bilboed.com>
-2007-09-03 Johan Dahlin <jdahlin@async.com.br>
+ * gst/playback/gstfactorylists.c:
+ playback: Fix the order in strcmp that I broke in previous commit.
- * gst/typefind/gsttypefindfunctions.c (plugin_init):
- Add an audio/x-nsf typefind function for the nsfdec element.
+2009-11-09 19:16:21 +0100 Edward Hervey <bilboed@bilboed.com>
-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
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Reduce number of calls to gst_type_find_peek.
+ Shaves off a couple percents off typefinding
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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()
+ * gst/playback/gstfactorylists.c:
+ playback: Avoid expensive API calls in tight loop.
+ We know we're dealing with GstPluginFeature.
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Add unit test for property settings
+ Also includes a regression test for bug #601104.
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- Fix typo.
+2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Never return a negative track number in get_uri()
- * 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.
+2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Don't set the track to 1 every time a device is set
+ Fixes bug #601104.
- * 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.
+2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstinputselector.c:
+ inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
- * 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.
+2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-31 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_sdes_copy_entry):
- Use g_strndup which does exactly what we want.
+2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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()
+ * 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.
-2007-08-30 Wim Taymans <wim.taymans@gmail.com>
+2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtcpbuffer.c:
- (gst_rtcp_packet_sdes_get_entry),
- (gst_rtcp_packet_sdes_copy_entry):
- * gst-libs/gst/rtp/gstrtcpbuffer.h:
- Fix and document SDES item data function.
- Add new function that makes a proper copy of SDES item data.
- API: gst_rtcp_packet_sdes_copy_entry()
+ * gst/playback/gstinputselector.c:
+ inputselector: Make sure that running_time->timestamp calculation never becomes negative
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
+2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
- * 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.
+ * 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.
-2007-08-30 Stefan Kost <ensonic@users.sf.net>
+2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Daniel Díaz <yosoy@danieldiaz.org>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Improve all-raw-caps detection for pads
- * configure.ac:
- * gst/Makefile.am:
- Check if libxml provides HTML parser which subparse needs.
- Fixes #451970.
+2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ basesrc: fix startup position in the ringbuffer
+ When we start and we need to produce the first sample, go to the next sample
+ that will be written into the ringbuffer instead of trying to go to sample 0.
+ We relied on rather small ringbuffer sizes to correctly go to the current
+ sample, which breaks whith large buffers.
+ Fixes #600945
- * ext/alsa/gstalsa.c:
- Fix typo and compilation on big endian systems.
+2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-29 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/subparse/gstssaparse.c:
- Convert SSA newline codes into actual newline characters (#470766).
+2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Synchronize video/text based on the running time
+ Instead of simply using the buffer timestamps.
- * 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).
+2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Clip text buffers to the text segment and reset segments properly
- * 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.
+2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Put the video segment into the instance struct instead of allocating it separately
- * 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.
+2009-11-06 09:05:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-26 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Check if text timestamp/duration is valid before clipping
- * gst-libs/gst/pbutils/missing-plugins.c:
- (gst_missing_plugin_message_get_installer_detail):
- Add missing separator in PID fallback case.
+2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/theoradec.c:
+ theoradec: printf format fix
- * 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.
+2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/audioconvert/Makefile.am:
- Add flags from -base before any more-remote dependencies.
+ * gst/gdp/gstgdpdepay.c:
+ gdpdepay: Clear adapter on flush and state change
+ Fixes #600469
-2007-08-23 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Based on a patch by: Davyd Madeley <davyd at madeley dot id dot au>
+ * gst/playback/gstinputselector.c:
+ inputselector: use _get_caps_reffed()
- * 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.
+2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-08-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ pad: rename new api from _refed to _reffed.
+ Due to popular demand rename the new api as we still can.
- * tests/examples/Makefile.am:
- Fix even more.
+2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2: avoid copying caps
+ Use get_caps_refed() when we can.
- * 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
+2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-23 Stefan Kost <ensonic@users.sf.net>
-
- * gst/volume/gstvolume.c:
- Move passthrough below gst_object_sync_values(). Fixes #442654.
+ * 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.
-2007-08-22 Wim Taymans <wim.taymans@gmail.com>
+2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- Clarify the docs a little.
+ * 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.
-2007-08-22 Stefan Kost <ensonic@users.sf.net>
+2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/volume/gstvolume.c:
- Enable liboil for float and add more details about problems with
- int16.
+ * 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.
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
- Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
+ * 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.
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst/playback/gstinputselector.c:
+ inputselector: also add inline to the proto to fix the build
+ Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
-2007-08-21 Wim Taymans <wim.taymans@gmail.com>
+2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
- (gst_ogg_demux_collect_chain_info):
- Also submit the eos page when trying to find the first timestamp.
- See #466717.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Initialize caps property with the default raw caps
-2007-08-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/audio.h:
- Use gst_util_uint64_scale() instead of doing the math
- with double for GST_FRAMES_TO_CLOCK_TIME() and
- GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
- prevents rounding errors. Fixes #467667.
+ * 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.
-2007-08-17 Wim Taymans <wim.taymans@gmail.com>
+2009-11-03 08:26:37 +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), (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()
+ * 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/rtsp/gstrtspdefs.h:
- Fix compilation with -pedantic.
- Add enum for _poll.
+2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
-2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- Patch by: Olivier Crete <tester at tester ca>
+2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * po/POTFILES.in:
+ po: queue2 has moved to core
-2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
- Only post buffering messages when we are a stream.
+ * 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.
-2007-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/pbutils/install-plugins.c:
- * gst-libs/gst/pbutils/missing-plugins.c:
- Small docs fix and addition.
+ * gst/playback/gstplaysink.c:
+ playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/icles/.cvsignore:
- * tests/icles/Makefile.am:
- * tests/icles/test-textoverlay.c:
- Add a dumb little test for textoverlay alignments.
+ * gst/playback/gstplaysink.c:
+ playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
+ Fixes bug #600027.
-2007-08-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Dan Williams <dcbw redhat com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
- * 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.
+2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-12 Wim Taymans <wim.taymans@gmail.com>
+ * docs/plugins/inspect/plugin-queue2.xml:
+ queue2: Remove inspect file
- * 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.
+2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-08-11 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/Makefile.am:
+ * gst/playback/gstqueue2.c:
+ queue2: Remove from gst-plugins-base
+ This is now in coreplugins.
- * 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)
+2009-10-28 11:29:36 +0200 Stefan Kost <ensonic@users.sf.net>
-2007-08-10 Stefan Kost <ensonic@users.sf.net>
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ docs: include more indexes
- * docs/libs/Makefile.am:
- Remove cruft and do some cleanups.
+2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/libs/gst-plugins-base-libs-docs.sgml:
- Prepare for comming gtkdoc features (rebase against online docs).
+ docs: turn entities into xi:includes
+ This is faster to process and easier to maintain. Its also less 80s.
-2007-08-10 Michael Smith <msmith@fluendo.com>
+2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Debug output fixes.
- * tests/check/elements/audiorate.c: (do_perfect_stream_test),
- (GST_START_TEST):
- Change the number of buffers used; 500 is too many and leads to
- timeouts.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: dump packets which we reject
-2007-08-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstqueue2.c:
- * gst/videorate/gstvideorate.c:
- Printf format fixes (#465028).
+ * tests/check/pipelines/.gitignore:
+ .gitignore: ignore basetime unit test binary
-2007-08-09 Michael Smith <msmith@fluendo.com>
+2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- If we have a large (> 1 second) discontinuity, push a series of
- smaller buffers rather than a single very large buffer. Avoids
- unreasonably large single buffer allocations when encountering a
- large gap.
- * tests/check/elements/audiorate.c: (GST_START_TEST),
- (audiorate_suite):
- Add a test for this.
+ * ext/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.
-2007-08-09 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_commit),
- (queue_remove_probe), (queue_threshold_reached):
+ * gst/playback/gstqueue2.c:
+ queue2: add custom acceptcaps function
- 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.
+2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Also, fix the spelling of Josep's surname in the ChangeLog.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement low/high watermark property
-2007-08-09 Stefan Kost <ensonic@users.sf.net>
+2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c:
- Do not leak oggmux instance.
-
- * ext/vorbis/vorbisenc.c:
- Also log values.
+ * tests/examples/seek/seek.c:
+ seek: add checkbox to enable buffering
-2007-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * po/hu.po:
- * po/it.po:
- * po/nl.po:
- * po/uk.po:
- * po/vi.po:
- Updated translations.
+ * 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).
-2007-08-08 Stefan Kost <ensonic@users.sf.net>
+2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- patch by: Yang Hong <hongyang@redflag-linux.com>
+ * gst/playback/gstplay-enum.c:
+ * gst/playback/gstplay-enum.h:
+ * gst/playback/gstplaybin2.c:
+ playbin2: add flag to enable decodebin buffering
+ Add a flag that enables buffering in decodebin.
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextoverlay.h:
- Add 'silent' property to GstTimeOverlay. Fixes #462979
+2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-08 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: buffering is implemented now
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/plugins/gst-plugins-base-plugins.args:
- * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
- (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
- (gst_uri_decode_bin_get_property), (gen_source_element):
- Add connection-speed property. Fixes #464690.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: buffering is implemented now
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: configure use-buffering on multiqueue
- * configure.ac:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect):
- Fix compilation on windows. Fixes #464320.
+2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-07 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: use 0 for max buffer size
- Patch by: Josep Torra Valles <josep@fluendo.com>
+2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (queue_threshold_reached),
- (gen_source_element), (setup_substreams),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (gst_play_base_bin_get_streaminfo_value_array):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_set_property), (gst_play_bin_get_property),
- (gst_play_bin_handle_redirect_message):
- Move connection-speed property from playbin to playbasebin so that we
- can also configure it in source elements that have the connection-speed
- property. Fixes #464028.
- Add some debug info here and there.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: set some reasonable defaults
-2007-08-06 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
- Properly respond to conversion queries. Fixes #464079.
+ * 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.
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
- (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
- (gst_audio_test_src_init_sine_table),
- (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Add float/double and int32 support to audiotestsrc. Fixes #460422.
- Also set the default volume to the default value specified in the
- GParamSpec.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add use-buffering property
+ Add a use-buffering property that will perform buffering on the parsed or
+ demuxed media.
-2007-08-03 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Jens Granseuer <jensgr at gmx dot net>
+ * 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/audioconvert/gstaudioquantize.c:
- Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
+2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: move error path down
- * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
- Add rdt manager for rdt transport.
- Fix parsing of RDT transport.
+2009-10-23 11:02:40 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: implement max queue size properties
- * configure.ac:
- Back to CVS
+2009-10-23 10:42:23 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.10.14 ===
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: add properties for buffering
+ Add properties that can be used to configure the multiqueue buffers and
+ buffering methods
-2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- releasing 0.10.14, "Light Years Ahead"
+ * 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
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/audio.c: (GST_START_TEST):
- Fix the test to reflect the behaviour of gst_audio_clip_buffer.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Don't destroy the suburidecodebin on errors
+ It can still be reused
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/audio.c:
- When clipping a buffer with no timestamp, assume it is
- within the segment without warnings.
+ * gst/playback/gstplaybin2.c:
+ playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
- Fixes: #460978
+2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-27 Wim Taymans <wim.taymans@gmail.com>
+ * 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.
- * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
- Fire the signal on the object, not the interface.
+2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * gst-libs/gst/rtsp/.cvsignore:
- Ber. Don't include the full path, idiot.
+2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
-2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/gsttheoraparse.h:
+ * ext/theora/theoraparse.c:
+ theora: Convert theoraparse to libtheora 1.0 API
- * gst-libs/gst/rtsp/.cvsignore:
- Ignore generated files.
+2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * 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.
+2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: move debug calculation into debug macro
+ Remove in_duration and move its calculation to
+ GST_LOG_OBJECT macro. This way it will only be calculated
+ if we have debug enabled.
- * gst-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.
+2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Removing unused variable
+ The in_stop variable was never read. Removing it.
- * gst-libs/gst/riff/Makefile.am:
- Pull in our dependencies from -base before those from outside.
+2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
-2007-07-26 Wim Taymans <wim.taymans@gmail.com>
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: be more accurate on offset math
+ Replace gst_util_uint64_scale_int for its rounding version
+ to improve accuracy and avoid inserting samples where
+ they aren't needed.
+ Fixes #499181
- * 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.
+2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Optimize a bit more
+ ...and add a FIXME for bug #598695 and explain
+ what we should do once Pango supports user fonts.
- * tests/check/libs/.cvsignore:
- Ignore the mixer test binary.
+2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
-2007-07-26 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/gstsubparse.h:
+ * tests/check/elements/subparse.c:
+ subparse: Add support for DKS subtitle format
+ Fixes bug #598936.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
- Gratuitous comment change to trigger a rebuild on the buildbots.
+2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Do shading as first operation
- * 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.
+2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-25 Wim Taymans <wim.taymans@gmail.com>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Only use a single cairo surface for drawing
+ ... and comment/optimize what is going on here a bit better.
- * 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>
+2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
- (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
- (gst_rtsp_message_init_response),
- (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
- (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
- (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
- (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
- (gst_rtsp_message_get_body), (dump_key_value):
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse):
- * gst-libs/gst/rtsp/gstrtsprange.h:
- * gst-libs/gst/rtsp/gstrtsptransport.c:
- * gst-libs/gst/rtsp/gstrtspurl.c:
- Fix some more RTSP docs.
- Add some missing methods for dealing with messages.
+ * gst/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.
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
- * gst-libs/gst/rtsp/gstrtspbase64.h:
- * gst-libs/gst/rtsp/gstrtspconnection.c:
- (gst_rtsp_connection_connect), (add_auth_header),
- (gst_rtsp_connection_write), (gst_rtsp_connection_send),
- (read_body), (gst_rtsp_connection_receive),
- (gst_rtsp_connection_next_timeout),
- (gst_rtsp_connection_reset_timeout),
- (gst_rtsp_connection_set_auth):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse):
- * gst-libs/gst/rtsp/gstrtspurl.h:
- Added beginnings of RTSP documentation.
+ * gst/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.
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/sdp/gstsdp.h:
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
- (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
- (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
- (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
- (gst_sdp_message_get_attribute_val),
- (gst_sdp_message_add_attribute), (gst_sdp_media_new),
- (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
- (gst_sdp_media_get_media), (gst_sdp_media_set_media),
- (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
- (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
- (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
- (gst_sdp_media_get_format), (gst_sdp_media_add_format),
- (gst_sdp_media_get_information), (gst_sdp_media_set_information),
- (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
- (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
- (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
- (gst_sdp_media_set_key), (gst_sdp_media_get_key),
- (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
- (gst_sdp_media_get_attribute_val_n),
- (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
- (print_media), (gst_sdp_message_dump):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- Document the SDP library.
- Add some of the missing SDPMedia methods.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix typo in previous mxf typefinder change
-2007-07-24 Wim Taymans <wim.taymans@gmail.com>
+2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/rtsp/Makefile.am:
- * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
- * gst-libs/gst/rtsp/gstrtspbase64.h:
- * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
- (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
- (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
- (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
- (parse_response_status), (parse_request_line), (parse_line),
- (gst_rtsp_connection_read), (read_body),
- (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
- (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
- (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
- (gst_rtsp_connection_set_auth):
- * gst-libs/gst/rtsp/gstrtspconnection.h:
- * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
- (gst_rtsp_strresult), (gst_rtsp_method_as_text),
- (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
- (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
- (gst_rtsp_find_method):
- * gst-libs/gst/rtsp/gstrtspdefs.h:
- * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
- (gst_rtsp_message_new), (gst_rtsp_message_init),
- (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
- (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
- (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
- (gst_rtsp_message_free), (gst_rtsp_message_add_header),
- (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
- (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
- (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
- (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
- (gst_rtsp_message_dump):
- * gst-libs/gst/rtsp/gstrtspmessage.h:
- * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
- (parse_npt_range), (parse_clock_range), (parse_smpte_range),
- (gst_rtsp_range_parse), (gst_rtsp_range_free):
- * gst-libs/gst/rtsp/gstrtsprange.h:
- * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
- (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
- (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
- (range_as_text), (rtsp_transport_mode_as_text),
- (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
- (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
- (gst_rtsp_transport_free):
- * gst-libs/gst/rtsp/gstrtsptransport.h:
- * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
- (gst_rtsp_url_free), (gst_rtsp_url_set_port),
- (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
- * gst-libs/gst/rtsp/gstrtspurl.h:
- * gst-libs/gst/sdp/Makefile.am:
- * gst-libs/gst/sdp/gstsdp.h:
- * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
- (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
- (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
- (gst_sdp_attribute_init), (gst_sdp_message_new),
- (gst_sdp_message_init), (gst_sdp_message_uninit),
- (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
- (gst_sdp_media_uninit), (gst_sdp_media_free),
- (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
- (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
- (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
- (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
- (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
- (gst_sdp_message_get_attribute_val),
- (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
- (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
- (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
- (gst_sdp_media_get_attribute_val_n),
- (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
- (read_string), (read_string_del), (gst_sdp_parse_line),
- (gst_sdp_message_parse_buffer), (print_media),
- (gst_sdp_message_dump):
- * gst-libs/gst/sdp/gstsdpmessage.h:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- Move SDP and RTSP from helper objects in -good to a reusable library.
- Use a proper gst_ namespace.
+ * gst/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 :)
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
- (vorbis_dec_flush_decode):
- Use the new buffer clipping function from gstaudio here.
+ * gst/playback/gstfactorylists.c:
+ decodebin2: avoid type checks
-2007-07-23 Sebastian Dröge <slomo@circular-chaos.org>
+2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
- * gst-libs/gst/audio/audio.h:
- * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
- API: Add buffer clipping function for raw audio buffers. Fixes #456656.
- Also add deprecation guards for gst_audio_structure_set_int() to the
- header.
+ * gst/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.
-2007-07-23 Stefan Kost <ensonic@users.sf.net>
+2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Cleanup the docs.
+ * gst/playback/gstdecodebin2.c:
+ gst/decodebin2: Don't expose chains if we're shutting down.
+ This avoids adding flushing pads to ourself
-2007-07-23 Wim Taymans <wim.taymans@gmail.com>
+2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Dan Williams <dcbw at redhat dot com>
+ * configure.ac:
+ * ext/pango/gsttextoverlay.c:
+ pango: bump pango requirement to stable version and remove ifdefs
+ Bump pango requirement from an ancient development version to an
+ ancient stable version.
- * gst/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.
+2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-07-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/.gitignore:
+ .gitignore: update after files got renamed
- * gst/playback/gsturidecodebin.c:
- Init debug category before using it.
+2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-07-21 Jan Schmidt <thaytan@noraisin.net>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppayload: small comment fix
- * 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.
+2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-07-20 Jan Schmidt <thaytan@noraisin.net>
+ * 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.
- * 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:
+2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
- Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
- Fixes: #152864
+ * 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.
- Add support for notifying mixer changes on the message bus, and
- implement it in alsamixer.
+2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
- API: gst_mixer_get_mixer_flags
- API: gst_mixer_message_parse_mute_toggled
- API: gst_mixer_message_parse_record_toggled
- API: gst_mixer_message_parse_volume_changed
- API: gst_mixer_message_parse_option_changed
- API: GstMixerMessageType
- API: GstMixerFlags
+ * gst-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.
-2007-07-20 Michael Smith <msmith@fluendo.com>
+2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 85d1530 to 0702fe1
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
- * tests/examples/volume/Makefile.am:
- * tests/icles/Makefile.am:
- We should use $(LIBM).
+ * ext/theora/theoradec.c:
+ theora: Make theoradec use gstvideo for image conversion
+ Vastly simplifies code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
-2007-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
- * tests/icles/Makefile.am:
- This needs -lm.
+ * 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
-2007-07-16 Wim Taymans <wim.taymans@gmail.com>
+2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
- * 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.
+ * 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
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
- Patch by: Bastien Nocera <hadess at hadess dot net>
+ * 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
- * 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.
+2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+ * 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
- Patch by: Jorn Baayen <jorn at openedhand dot com>
+2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
- * 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
+ * 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
-2007-07-14 Wim Taymans <wim.taymans@gmail.com>
+2009-09-09 21:57:08 +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.
+ * 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
-2007-07-13 Wim Taymans <wim.taymans@gmail.com>
+2009-09-09 21:50:57 +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.
+ * 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
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-09 21:48:08 +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.
-
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Wim Taymans <wim.taymans@gmail.com>
-
- * 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 Wim Taymans <wim.taymans@gmail.com>
-
- * 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.
+ * 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
-2007-07-12 Stefan Kost <ensonic@users.sf.net>
+2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
- * configure.ac:
- Use pkg-config to locate check.
+ * 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
-2007-07-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
* configure.ac:
- * tests/check/elements/volume.c: (GST_START_TEST):
- Fix 'make check' build against core CVS.
-
-2007-07-10 Stefan Kost <ensonic@users.sf.net>
-
- * gst-libs/gst/interfaces/propertyprobe.c:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Make gtk-doc happy.
-
-2007-07-08 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theora: remove support for outdated granulepos hack
+ This is in preparation to switching to switching to the new Theora API
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * gst-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.
+2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
-2007-07-06 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
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Fix location of includes in the docs.
+2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
-2007-07-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ theora: Ignore the center property, always set video to top left
+ This is not a necessary property, the output will be identical no matter
+ what.
+ https://bugzilla.gnome.org/show_bug.cgi?id=594729
- * gst/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
+2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-07-05 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
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Simplify --extra-dir as gtkdoc scans recursively.
+2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-07-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post a element message on the bus with the stream topology
+ Fixes bug #598533.
- * 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.
+2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-29 Wim Taymans <wim.taymans@gmail.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Store the "endcaps" of a chain
+ This are the caps that either resulted in a deadend if
+ no plugin for them could be found or raw caps.
- * gst/audioconvert/audioconvert.c:
- Include math.h to fix compilation.
+2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-29 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Store for every chain, which pad resulted in its creation
- * 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
+2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-06-28 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
- * 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.
+2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
- Add unit test for the above.
+ * common:
+ Automatic update of common submodule
+ From a3e3ce4 to 85d1530
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
- Add description for Windows Media RTP caps.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
- * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
- Remove RTP fields that don't define the format from caps.
+2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
-2007-06-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: fix timestamp comparission, Fixes #597407
- * 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.
+2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
* 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.
-
-2007-06-27 Wim Taymans <wim@fluendo.com>
-
- * 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 Stefan Kost <ensonic@users.sf.net>
-
- * 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.
-
-2007-06-25 Tim-Philipp Müller <tim at centricular dot net>
-
- * po/POTFILES.in:
- Add more files with translatable strings (#450875).
+ * 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.
-2007-06-23 Edward Hervey <edward@fluendo.com>
+2009-10-08 19:55:42 +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.
+ * 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.
-2007-06-22 Jan Schmidt <thaytan@noraisin.net>
+2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * MAINTAINERS:
- Updating all the maintainers files
+ * 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.
-2007-06-21 Stefan Kost <ensonic@users.sf.net>
+2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/audioconvert/gstaudioconvert.c:
+ audioconvert: track active conversion in perf log
-2007-06-20 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
- * 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/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
-2007-06-19 Andy Wingo <wingo@pobox.com>
+2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_init): Enable pull-mode operation.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: whitespace fixes
-2007-06-19 Michael Smith <msmith@fluendo.com>
+2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Change minimum rate back to 1000 to allow low-sample-rate wav files
- to play back.
+ * ext/theora/theoradec.c:
+ theoradec: avoid confusing error
-2007-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
- * po/vi.po:
- Update translations.
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ vorbis: fixes warings in macosx snow leopard
-2007-06-15 David Schleef <ds@schleef.org>
+2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst/playback/gstqueue2.c:
- Fix compile error from ignored return value.
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraparse.c:
+ theora: fixes warnings on macosx snow leopard
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.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.
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ ogg: fixes warnings on macosx snow leopard
-2007-06-15 Michael Smith <msmith@fluendo.com>
+2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.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.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: fix a warning in macosx
-2007-06-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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-libs/gst/tag/tags.c:
+ tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
-2007-06-13 Edward Hervey <edward@fluendo.com>
+2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Fix build on MacOSX.
+ * tests/check/gst-plugins-base.supp:
+ check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
-2007-06-13 Wim Taymans <wim@fluendo.com>
+2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
- * 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.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ audioconvert: change the format instead of cast as ensonic asked
-2007-06-12 Wim Taymans <wim@fluendo.com>
+2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
- Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
+ redo of valid part of my previous revert.
- * 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.
+2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
-2007-06-11 Wim Taymans <wim@fluendo.com>
+ * 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.
- Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.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.
+ * 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'
-2007-06-07 Sebastien Moutte <sebastien@moutte.net>
+2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
- * 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.
+ * gst/videorate/gstvideorate.c:
+ videorate: fix warning in macosx
-2007-06-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
- The BOS page of the first Dirac video stream needs to come before
- the BOS page of any Vorbis streams or other audio streams, just like
- it is with Theora.
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: fix warning in macosx
-2007-06-07 Wim Taymans <wim@fluendo.com>
+2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
- * gst/playback/gstqueue2.c: (gst_queue_get_range):
- Fix compilation.
+ * common:
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: fixes warning: format not a string literal and no format arguments
-2007-06-06 Wim Taymans <wim@fluendo.com>
+2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
- Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ audio: fix warnings building on macosx
- * 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.
+2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-06-06 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: chwck formats just once per _chain()
- * configure.ac:
- * gst/playback/gstqueue2.c: (gst_queue_create_read):
- Some more fseeko checks.
+2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-06-06 Wim Taymans <wim@fluendo.com>
+ * 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.
- * configure.ac:
- check for large file support.
+2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-06-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 19fa4f3 to a3e3ce4
- Based on a patch by Sven Arvidsson <sa at whiz dot se>:
+2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
- * gst/subparse/gstsubparse.c: (parse_subrip),
- (subviewer_unescape_newlines), (parse_subviewer),
- (gst_sub_parse_data_format_autodetect),
- (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
- * gst/subparse/gstsubparse.h:
- Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add a unit test for both SubViewer formats.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Fix type-punning warning
-2007-06-01 Michael Smith <msmith@fluendo.com>
+2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
- Don't overflow intermediate values when seeking to large time values
- in audiotestsrc.
+ * gst/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
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_have_data),
- (gst_queue_create_read), (gst_queue_read_item_from_file),
- (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
- Include stdio to define fseeko.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
-2007-06-05 Wim Taymans <wim@fluendo.com>
+2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Edward Hervey <edward@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Don't post missing plugin messages twice
+ decodebin2 already posts them after emitting the unknown-type signal,
+ there's no need to post another one.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
- (gst_v4lsrc_query):
- Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
+2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-06-05 Tim-Philipp Müller <tim at centricular dot 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.
- * 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.
+2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-06-05 Wim Taymans <wim@fluendo.com>
+ * sys/ximage/ximagesink.c:
+ ximagesink: only start event thread if needed
+ The event thread is doing 20 wakeups per second to poll the events. If one
+ runs ximagesink with handle-events=false and handle-expose=false then we can
+ avoid the extra thread.
- * 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.
+2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-06-05 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/check/libs/video.c:
+ tests/check/libs/video.c: Update strides for Y41B
- * configure.ac:
- Back to CVS
+2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.10.13 ===
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: we can use GLib 2.18 API unconditionally now
-2007-06-05 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
- releasing 0.10.13, "What's Going on?"
-
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
+ configure: bump GLib requirement to 2.18
+ Bump required GLib version as per the release planning docs.
- * 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
+2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Wim Taymans <wim@fluendo.com>
- Patch by: Sebastian Dröge <slomo@circular-chaos.org>
-
-2007-05-31 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst-libs/gst/floatcast/floatcast.h:
- Define inline when needed on win32 builds. Fixes: #441295
- Patch by: Sebastien Moutte <sebastien@moutte.net>
-
-2007-05-29 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (queue_overrun),
- (no_more_pads_full):
- Stop buffering when the group is commited because the queues filled up.
- Fixes #442024.
+ * gst-libs/gst/interfaces/tuner.c:
+ docs: clarify GstTuner docs in two places
-2007-05-25 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2007-05-24 Wim Taymans <wim@fluendo.com>
+2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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
-2007-05-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
- * 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
+ * gst-libs/gst/video/video.h:
+ Update Since tags for NV12/NV21
+ They are added in 0.10.26 now, not 0.10.25
-2007-05-24 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-23 15:31:50 +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
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
-2007-05-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-23 11:03:57 +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.
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-23 10:25:02 +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.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ Add NV12 and NV21 formats
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
- Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * 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
- * 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.
+2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
-2007-05-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ [videotestsrc] Fix Y41B
+ Chroma components should be aligned on 4byte boundaries.
+ https://bugzilla.gnome.org/show_bug.cgi?id=595849
- * ext/cdparanoia/gstcdparanoiasrc.c:
- Repair umlaut.
+2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ streamvolume: Define cbrt() if it's not available
+ Fixes build on Win32, bug #597537.
- * 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.
+2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstfactorylists.c:
+ factorylist: Use gst_caps_can_intersect() instead of _intersect()
+ This is faster and results in less allocations.
- * 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.
+2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-21 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Don't set the external ghostpads blocked but only their targets
+ Pad blocks should never be done on external pads as outside elements
+ might want to use their own pad blocks on them and this will lead to
+ conflicts and deadlocks.
- * ext/theora/theoradec.c: (theora_handle_data_packet):
- Don't look inside 0-length packets (which indicate duplicated
- frames)
+2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-05-21 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Only use the object lock for protecting the subtitle elements
+ Using the decodebin lock will result in deadlocks if the subtitle encoding
+ is accessed from a pad-added handler.
- * ext/cdparanoia/gstcdparanoiasrc.c:
- (gst_cd_paranoia_src_read_sector):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Small cleanups.
+2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix typo.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Improve debugging of pad blocks
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_set_gst_timestamp):
- Add some FIXME
+2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (queue_underrun_cb):
- And some debug info when a FIXME path is hit.
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
-2007-05-21 Wim Taymans <wim@fluendo.com>
+2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
- * 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.
+ * configure.ac:
+ configure: Add an 'else' to pangocairo check
+ Otherwise it exits if it fails.
- * 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.
+2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_change_state):
- Fix typo in comment.
+2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@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.
+ * 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
- * gst/volume/Makefile.am:
- Fix tabs.
+2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
-2007-05-18 Stefan Kost <ensonic@users.sf.net>
+ * 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
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
- set_option, get_option, _gst_reserved):
- Revert reordering functions (keep ABI).
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ ffmpegcolorspace: Use the ffmpegcolorspace debug category
+ Move gstffmpegcodecmap debug to the ffmpegcolorspace category
-2007-05-17 Jan Schmidt <thaytan@mad.scientist.com>
+2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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/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.
- Fixes: #393975
+2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
- Clean up the X event handling loop and make them the same for
- both xvimagesink and ximagesink while I'm at it.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: detect 'ftypqt ' as video/quicktime
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
- Make decodebin2 autoplug depayloaders too.
+ * configure.ac:
+ back to development -> 0.10.25.1
- * gst/playback/gsturidecodebin.c: (source_new_pad):
- Set the newly created decoder in a usable state when autoplugging a
- dynamic source such as RTSP.
+=== release 0.10.25 ===
-2007-05-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-10-05 13:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ Release 0.10.25
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
- * gst/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>
+2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
* 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.
+ * 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
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstqueue2.c: (gst_queue_get_type),
- (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
- (apply_segment), (apply_buffer), (update_buffering),
- (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_handle_sink_event), (gst_queue_is_filled),
- (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
- (plugin_init):
- fix build.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
-2007-05-17 Wim Taymans <wim@fluendo.com>
+2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/Makefile.am:
- * gst/playback/gstqueue2.c: (gst_queue_get_type),
- (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
- (gst_queue_getcaps), (gst_queue_bufferalloc),
- (gst_queue_acceptcaps), (update_time_level), (apply_segment),
- (apply_buffer), (update_buffering), (reset_rate_timer),
- (update_rates), (gst_queue_locked_flush),
- (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
- (gst_queue_handle_sink_event), (gst_queue_is_empty),
- (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
- (gst_queue_loop), (gst_queue_handle_src_event),
- (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
- (gst_queue_src_activate_push), (gst_queue_change_state),
- (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
- On our way to playbin2 this is the new network queue that does buffering
- all by itself using high and low watermarks. It can also measure up and
- downstream bandwidth to optimally size the queue.
-
-2007-05-17 Michael Smith <msmith@fluendo.com>
-
- * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
- Use the segment->last_stop value to calculate the next timestamp to
- generate after a seek; not the segment->start value.
+ * gst/playback/gstplaysink.c:
+ playsink: make the lock recursive for now
+ Fixes #583255
-2007-05-15 David Schleef <ds@schleef.org>
+2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/Makefile.am: Install docs even when --disable-gtk-doc
- is disabled. This matches the behavior of gtk+. Fixes #349099.
+ * gst/playback/gstplaysink.c:
+ playsink: fix the vis property getter
-2007-05-15 Wim Taymans <wim@fluendo.com>
+2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
- Some more chained streaming ogg timestamp fixes.
+ * gst-plugins-base.spec.in:
+ Add missing file to spec file
-2007-05-15 Wim Taymans <wim@fluendo.com>
+2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_handle_page):
- Add some FIXMEs.
- Fix chain start/stop segment handling based on patch by
- <ahalda at cs dot mcgill dot ca> see #320984.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
-2007-05-15 Michael Smith <msmith@fluendo.com>
+2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
- * configure.ac:
- We don't require a C++ compiler. So don't require one.
+ * 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.
-2007-05-15 Stefan Kost <ensonic@users.sf.net>
+2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.3 pre-release
-2007-05-15 Stefan Kost <ensonic@users.sf.net>
+2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
- patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: don't ever return NULL in list of strings.
- * 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
+2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-14 David Schleef <ds@schleef.org>
+ * gst/playback/gstplaysink.c:
+ playsink: Expose mute,volume,vis-plugin and font-desc properties
+ https://bugzilla.gnome.org/show_bug.cgi?id=594623
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Add support for video/x-raw-bayer.
+2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-12 David Schleef <ds@schleef.org>
+ * gst/playback/gstplaysink.c:
+ GstPlaySink: Expose 'reconfigure' as an action signal.
- * sys/xvimage/xvimagesink.c:
- Add some sanity checking for the XVImage size returned by X.
- Related to #377400.
+2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-12 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ GstPlaySink: Expose flags as a gobject property.
- * 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.
+2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-05-12 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-05-11 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * 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):
+2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
- 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.
+ * 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.
- Also cleanup the MPEG elementary and MPEG-TS detection functions a
- little.
+2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Tested against my media test directory, with some improvements and
- no regressions.
+ * 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.
-2007-05-10 Wim Taymans <wim@fluendo.com>
+2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
- (queue_out_of_data):
- Connect to the new queue "pushing" signal instead of the broken
- "running" one.
+ * ext/theora/theoraenc.c:
+ theoraenc: Fix a string leak in _getcaps()
-2007-05-09 Sebastien Moutte <sebastien@moutte.net>
+2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
-2007-05-09 Michael Smith <msmith@fluendo.com>
+2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- * ext/theora/theoradec.c: (theora_handle_type_packet):
- gst_pad_push_event() does not return a GstFlowReturn!
+ * 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.
-2007-05-09 Wim Taymans <wim@fluendo.com>
+2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
- * tests/examples/seek/scrubby.c: (stop_cb), (main):
- * tests/examples/seek/seek.c: (do_seek):
- Some small cosmetic changes.
+ * 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.
-2007-05-08 Stefan Kost <ensonic@users.sf.net>
+2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
- gst_adder_change_state):
- * gst/adder/gstadder.h (bps, offset, collect_event, segment,
- segment_pending, segment_position, segment_rate):
- Handle playback-rate on adder.
+ * gst/playback/gstplaybin2.c:
+ playbin2: Implement GstStreamVolume interface
-2007-05-07 Michael Smith <msmith@fluendo.com>
+2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (gst_theora_dec_reset),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_type_packet), (theora_dec_change_state):
- Don't push events (newsegment, tags) before initialising the
- decoder.
- This is neccesary for seeking to work correctly in gnonlin.
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c:
+ volume: Implement GstStreamVolume interface
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst/adder/gstadder.c:
- * gst/audiotestsrc/gstaudiotestsrc.c
- (gst_audio_test_src_create_white_noise):
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
- VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
- volume_sink_template, volume_src_template, gst_volume_init,
- volume_process_double, volume_process_int16,
- volume_process_int16_clamp):
- Doc fixes and formatting.
+ * 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.
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
- Minimal check for volume's GstController usability; also another
- test for #422295.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: properly fix the HTTP manual mode
+ When we're not parsing HTTP, return EPARSE when we get an HTTP
+ message.
-2007-05-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track):
- Fix it so that it (a) makes sense and (b) doesn't break
- everything cdda-related including the unit test.
+ * gst-libs/gst/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
-2007-05-04 Stefan Kost <ensonic@users.sf.net>
+2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_add_track):
- Fix build when disabling asserts.
+ * 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.
-2007-05-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
- When XShm is not available, we might get row strides that are not
- rounded up to multiples of four; this is bad, because virtually
- every RGB-processing element in GStreamer assumes rowstrides are
- rounded up to multiples of four, so let's allocate at least enough
- memory to avoid crashes in this case. The image will still be
- displayed distorted though if this happens, so that still needs
- fixing (maybe by allocating a bigger image with an 'even' width
- and then clipping it appropriately when rendering - something for
- Xlib aficionados in any case).
+ * gst/videorate/gstvideorate.c:
+ videorate: Add Since marker for the new skip-to-first property
-2007-05-03 Michael Smith <msmith@fluendo.com>
+2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- If a buffer doesn't have a timestamp, assume it's contiguous with
- the previous buffer, and synthesise timestamps appropriately.
+ * gst/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.
-2007-05-03 Edward Hervey <edward@fluendo.com>
+2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/videorate.c: (GST_START_TEST):
- Set buffer timestamp to a valid value in order to test the buffer
- really does stay in videorate.
+ * gst-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
-2007-05-03 Edward Hervey <edward@fluendo.com>
+2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2007-05-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
- * gst/playback/gstdecodebin2.c: (plugin_init):
- Better error message for text files.
+ * 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
-2007-04-29 Wim Taymans <wim@fluendo.com>
+2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
- Fix offset bug in generation RR packets.
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ playbin(2): Document that the volume property uses a linear scale
+ Fixes bug #571610.
-2007-04-27 Julien MOUTTE <julien@moutte.net>
+2009-09-10 14:04:53 +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/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.
-2007-04-27 Wim Taymans <wim@fluendo.com>
+2009-09-10 12:38:16 +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/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.
-2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
- * 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/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
-2007-04-25 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.
+2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
-2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
+ * 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: Christian Kirbach <Christian dot Kirbach at googlemail com>
+2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
- * sys/ximage/ximagesink.c:
- Fix build if XShm is not available (#432362).
+ * 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.
-2007-04-24 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
- Initalize the AudioConvertCtx with zeroes, otherwise it will contain
- pointers to random memory which are passed to g_free() when
- audio_convert_prepare_context() is called the first time.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle kate and cmml as sparse streams too
-2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Dan Williams <dcbw redhat com>
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: Better handling of sparse streams by sending segment updates
+ Fixes bug #397419.
- * 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.
+2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-04-23 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gsturidecodebin.c:
+ docs: tell a biit more about uri-decodebin and buffering
- * 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
+2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: take clock time in setcaps
+ Take the time of the clock so that the last_time field is set. This is important
+ for sinks that restart their internal ringbuffer after a caps change and need to
+ know the last know position.
- Patch by: Olivier Crete <tester at tester ca>
+2009-09-09 18:24:15 +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).
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: add some more debug
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@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.
+ * ext/theora/theoraenc.c:
+ theoraenc: Print a debug message with supported formats
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
- Patch by: Zeeshan Ali <zeenix gmail com>
+ * 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.
- * 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).
+2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theora: Implement support in theoraenc for Y444 and Y42B
+ Fixes bug #594165.
- * gst/audioresample/gstaudioresample.c:
- Make more functions static, just because we can.
+2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
-2007-04-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theora: Refactor the buffer copy code
- * tests/check/elements/audioresample.c:
- Add unit test for audioresample shutdown crasher (#420106).
+2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
-2007-04-20 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoraenc.c:
+ theora: Split yuv_buffer creation into its own function
- * gst/subparse/gstsubparse.c:
- * gst/subparse/samiparse.c:
- Use GST_DISABLE_XML here
+2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_navigation_send_event):
- * sys/xvimage/xvimagesink.h:
- Include stdlib.h when using atoi.
-
- * tests/check/elements/playbin.c: (playbin_suite):
- Use GST_DISABLE_REGISTRY here
+ * ext/theora/theoraenc.c:
+ theora: Split out buffer resize in its own function
-2007-04-19 Michael Smith <msmith@fluendo.com>
+2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
- * 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/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.
-2007-04-18 Stefan Kost <ensonic@users.sf.net>
+2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/pipelines/.cvsignore:
- Fix build.
+ * tests/examples/seek/seek.c:
+ seek: make stop state configurable
+ Make it easy to experiment with different stop states (NULL and READY)
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Allow random depths between 1 and 32 instead of only multiplies of 8.
+ * 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
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ baseaudiosink: whitespace fixes
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Correctly support 4, 6 and 8 channels with normal PCM and float
- wav files.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add more debug
- 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.
+2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Add support for float, alaw and mulaw in extensible wav files.
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ whitespace fixes
- 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.
+2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
- Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
- * 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.
+2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/video/gstvideosink.c:
+ * gst-libs/gst/video/gstvideosink.h:
+ video: add GstVideoSinkClass::show_frame()
+ Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
+ vfuncs and add some gtk-doc chunks.
+ API: GstVideoSinkClass::show_frame()
- * ext/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.
+2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * 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.
+2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix compiler warning with MSVC
+ Fixes bug #594275.
- * 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
+2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
-2007-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: fix event forwarding
- * 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.
+2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
-2007-04-13 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
+ Fixes #594258
- * ext/theora/theoradec.c: (theora_dec_sink_event):
- Fix misleading comment.
+2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-04-13 Stefan Kost <ensonic@users.sf.net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ fix whitespace
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- More sanity checks for the header fields.
+2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+ * 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-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.
+2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-04-12 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: disable typefinder for headerless flac
+ Disable headerless flac typefinder as long as it happily typefinds anything
+ including /dev/urandom as flac and as long as it's not particularly useful
+ given that such streams don't really exist in the wild.
+ Also fix up some comments so that gtk-doc doesn't complain about them.
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
- (gst_video_rate_chain):
- Add some debug.
+2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
- * tests/check/elements/videorate.c: (GST_START_TEST),
- (videorate_suite):
- Added check for videorate changing caps handling. Closes #421834.
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix small memory leak when setting window title
-2007-04-12 Michael Smith <msmith@fluendo.com>
+2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Use scale functions to avoid overflow when calculating duration of
- vorbis buffers.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix small memory leak when setting window title
-2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
- API: add gst_tag_freeform_string_to_utf8() (#405072).
+ * .gitignore:
+ introspection: Add *.gir and *.typelib to .gitignore
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
- Use gst_tag_freeform_string_to_utf8() here.
+2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-12 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
- * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
- (gst_gdp_pay_sink_event):
- Make sure we set the IN_CAPS flag correctly.
+2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
- Get the IN_CAPS flag before we call functions that mess with the flags.
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Fix introspection build by ordering sources/headers in dependency order
-2007-04-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
- gst_gdp_pay_chain, gst_gdp_pay_sink_event):
- Only stamp buffers with offset/offset_end right before they get
- pushed. This ensures offset continuity, which was not the case
- before as shown by
- gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Remove debug echo
-2007-04-06 Wim Taymans <wim@fluendo.com>
+2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin.c: (add_sink),
- (gst_play_bin_change_state):
- Activate sync in playbin, we are ready to handle it for live streams.
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix build of introspection data by using dependency order for the headers/sources
-2007-04-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/playbin.c:
- (test_sink_usage_video_only_stream), (playbin_suite):
- Add small test for stream-info-value-array code paths.
+ * gst-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
-2007-04-05 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_skew_slaving):
- Don't try to create invalid calibration parameters by making the
- internal time go backwards, instead make external time go forward.
+ * gst-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
-2007-04-05 Wim Taymans <wim@fluendo.com>
+2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst-libs/gst/video/Makefile.am:
+ video: Add gobject-introspection support
- * gst/playback/gstplaybasebin.c: (add_stream):
- Fix leak in add_stream(), when g_value_set_object() increases the
- refcount of streaminfo object. Fixes #426250.
+2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-03 David Schleef <ds@schleef.org>
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Add gobject-introspection support
- * 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.
+2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-04-03 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: Add gobject-introspection support
- Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
- (deactivate_free_recursive):
- Decodebin2 doesn't unref pads it obtains in some occasions:
- - multiqueue src pads, when either connecting further or exposing
- - sink pads of new autoplugged elements
- - peer pads when recursively freeing elements
- Fixes #425455.
+ * gst-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
-2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Add audio/x-raw-float support, now that audioconvert support
- non-native endianness floats.
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Add gobject-introspection support
-2007-03-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/rtp/Makefile.am:
+ rtp: Add gobject-introspection support
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler dot de>
- with some minor changes
+ * gst-libs/gst/riff/Makefile.am:
+ riff: Add gobject-introspection support
- * 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.
+2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: Add gobject-introspection support
- * 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.
+2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/netbuffer/Makefile.am:
+ netbuffer: Add gobject-introspection support
- * 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.
+2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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/interfaces/Makefile.am:
+ interfaces: Add gobject-introspection support
-2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
+2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- PCM samples with width=8 must be always unsigned, no matter what
- depth they have.
+ * gst-libs/gst/fft/Makefile.am:
+ fft: Add gobject-introspection support
-2007-03-29 Andy Wingo <wingo@pobox.com>
+2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
- perfect offsets also, not just timestamps.
+ * gst-libs/gst/cdda/Makefile.am:
+ cdda: Add gobject-introspection support
+ This is disabled for now until gobject-introspection is fixed
- * tests/check/elements/videorate.c (test_more): Test that given
- any incoming offsets, that videorate produces perfect offsets.
+2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Add gobject-introspection support
- * gst-libs/gst/riff/riff-ids.h:
- Add some more RIFF formats.
+2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-29 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * gst-libs/gst/app/Makefile.am:
+ app: Add gobject-introspection support
- * 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.
+2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-27 Sebastian Dröge <slomo@circular-chaos.org>
+ * common:
+ Automatic update of common submodule
+ From 00a859e to 19fa4f3
- * 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.
+2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-27 Tim-Philipp Müller <tim at centricular dot 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/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.
+2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-26 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
- Patch by: René Stadler <mail at renestadler de>
+2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- When writing out floating-point numbers to vorbis comment tags, always
- use the same character as separator no matter what the current locale is
- (fixes #423051).
+ * gst/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.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit tests for replaygain tags in vorbis comments (closes #423055).
+2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: add some debugging
- * 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.
+2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-26 Jan Schmidt <thaytan@mad.scientist.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-libs/gst/rtp/Makefile.am:
- The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
+2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-23 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix frame duration calculations
+ Fix the calculation of the frame duration and rtp timestamps.
+ Add some debugging
- * gst/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.
+2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppay: add some debugging
- * 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.
+2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-22 Wim Taymans <wim@fluendo.com>
+ * 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/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.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: move function around
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c: (add_sink):
- Use GST_PTR_FORMAT to log caps.
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix sample duration calculation
-2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Young-Ho Cha <ganadist at chollian net>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: more refactoring
+ Unify the sample/frame buffer handling code by making the functions plugable.
- * gst/subparse/samiparse.c: (handle_start_font):
- Special-case some more colour names that pango doesn't handle by
- default. Fixes #420578.
+2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-20 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-19 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ * win32/common/libgstrtp.def:
+ audiortppayload: refactor and cleanup
+ Always use the adapter when we need to fragment the incomming buffer. Use more
+ modern adapter functions to avoid malloc and memcpy. The overall result is that
+ the code looks cleaner while it should be equally fast and in some case avoid a
+ memcpy and malloc.
+ Use the adapter timestamping functions for more precise timestamps in case of
+ weird disconts.
+ Cache some values instead of recalculating them.
+ Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
+ the internal adapter.
+ API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
- * gst/playback/gstdecodebin.c: (remove_fakesink):
- Don't post STATE_DIRTY anymore.
+2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * common:
+ Update common
-2007-03-17 David Schleef <ds@schleef.org>
+2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * REQUIREMENTS: Update this file, change the formatting to make
- it more consistent, plus more machine readable.
+ * 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
-2007-03-16 Michael Smith <msmith@fluendo.com>
+2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2007-03-16 Michael Smith <msmith@fluendo.com>
+2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
-
-2007-03-15 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppay: add macro to cast
- * 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.
+2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: code cleanups
- Patch by: Olivier Crete <olivier.crete@collabora.co.uk>
+2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@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.
+ audiortppayload: don't check adapter
+ the adapter is never NULL so we don't need to check it.
+ Use _scale functions to avoid overflows.
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- Update translations from translation project.
+ * 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.
-2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@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/subparse/gstsubparse.c:
+ subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
-2007-03-13 Michael Smith <msmith@fluendo.com>
+2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@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...
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: fix timestamping for when we do not have a clock yet
+ Should fix #559049.
-2007-03-10 Sebastien Moutte <sebastien@moutte.net>
+2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
- Use gst_guint64_to_gdouble for conversion.
- * win32/MANIFEST:
- Add new files to the win32 MANIFEST.
- * win32/common/libgstaudio.def:
- * win32/common/libgstpbutils.def:
- Add new exported functions.
- * win32/vs6/gst_plugins_base.dsw:
- * win32/vs6/libgstdecodebin.dsp:
- * win32/vs6/libgstplaybin.dsp:
- Change the link to libgstpbutils.lib.
- * win32/vs6/libgstdecodebin2.dsp:
- Add a new project for decodebin2.
- * win32/vs6/libgstpbutils.dsp:
- Add a new project for pbutils.
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: don't log not-yet-initialised integer value
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-09-01 14:28:48 +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).
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
+ And reflow code to be more indent friendly.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Add unit test for the above.
+2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * tests/check/elements/subparse.c: (GST_START_TEST),
- (subparse_suite):
- Add unit test for MPL2 subtitle format (#413799).
+2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- Patch by: Kamil Pawlowski <kamilpe gmail com>
+2009-08-31 11:10:55 +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/playback/gstdecodebin2.c:
+ decodebin2: Post missing plugin messages before any error messages
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- We require core CVS for the new buffer metadata copy functions.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: safely handle the indexes
-2007-03-09 Wim Taymans <wim@fluendo.com>
+2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/tag/gstid3tag.c:
- Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
- Fixes #414496.
+ * win32/common/libgstrtsp.def:
+ def: add new rtsp symbols
- Patch by: Alex Lancaster <alexl at users sourceforge net>
+2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: whitespace fixes.
- * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
- (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
- Improve adapter usage and comments.
+2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
-2007-03-09 Wim Taymans <wim@fluendo.com>
+ * gst/gdp/gstgdppay.c:
+ Bug 593035 - set IN_CAPS for streamheader buffer
- * 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.
+2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform):
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
- Basetransform copied the metadata for us.
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: The internally linked pad of the selector might be NULL in some cases
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
- (gst_text_overlay_video_event):
- Some more logging. Only accept newsegment events in TIME format and
- send a WARNING message if they are not in TIME format.
+ * gst/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.
- * 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.
+2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Michael Smith <msmith@fluendo.com>
+ * 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.
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
- Replace AC3 typefinder with one that isn't terrible, and actually
- works usefully.
+2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Detect AVF files as RIFF files too
+ AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
+ Partially fixes bug #593117.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- fix error category and translatable string
-
+2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/audioresample.c:
+ audioresample: Add unit test for checking for timestamp drifts
+ This also checks for perfect timestamping and offsetting.
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Fix up utils => pbutils here too.
+2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-09 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.
- * 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.
+2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * tests/check/elements/audioresample.c:
+ audioresample: Improve debugging a bit in the unit test
- * 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)
+2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * tests/check/Makefile.am:
- Don't test libvisual elements in the states check, because libvisual
- seems to leak internally.
+2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
- Re-enable the alsa and states tests now that there's new suppressions
- in gst.supp.
+ * configure.ac:
+ * gst/subparse/gstsubparse.c:
+ subparse: Remove dependency on regex.h as it's not used anyway
+ Fixes bug #592544.
- * tests/check/elements/alsa.c: (GST_START_TEST):
- Don't leak the alsamixer we instantiated.
+2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer overflow when pushing the drain
- * 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.
+2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: Fix timestamp drift
+ Fixes bug #591934.
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Extra log line.
+2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
- * 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)
+ * 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/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.
+2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-03-07 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * 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).
+2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
- * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Set "untranslated-label" when constructing mixer track objects.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Mark Transport as supporting multiple values.
- * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
- Unit test to check the above.
+2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-07 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added missing Since tags.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
- Fix confusing debug message.
+2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.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-plugins-base.doap:
- update doap file with new version
+2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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>
- * configure.ac:
- Back to CVS
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made gst_rtsp_watch_queue_data() thread safe.
-=== release 0.10.12 ===
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_connection_set_http_mode().
+ With gst_rtsp_connection_set_http_mode() it is possible to tell the
+ connection whether to allow HTTP messages to be supported. By enabling HTTP
+ support the automatic HTTP tunnel support will also be disabled.
+ API: gst_rtsp_connection_set_http_mode()
- * configure.ac:
- releasing 0.10.12, "Zombie Horde"
+2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * configure.ac:
- Bump version to 0.10.11.4 pre-release
+2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-06 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/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- Fix regression that made GStreamer skip the first samples of audio.
- Fixes #414684.
+2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-05 Jan Schmidt <thaytan@mad.scientist.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.
- * configure.ac:
- Bump version to 0.10.11.3 pre-release
+2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-05 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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).
- * po/POTFILES.in:
- Update paths for the rename from utils to pbutils to fix the build.
+2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Do not split headers which should not be split.
+ From RFC 2068 section 4.2: "Multiple message-header fields with the same
+ field-name may be present in a message if and only if the entire
+ field-value for that header field is defined as a comma-separated list
+ [i.e., #(values)]." This means that we should not split other headers which
+ may contain a comma, e.g., Range and Date.
- * gst-libs/gst/pbutils/Makefile.am:
- Change directory to install headers in from gst/utils to gst/pbutils
- as well.
+2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * 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
+2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-02 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improve parse_line().
+ Make parse_line() handle keys with multiple values on one line correctly.
- * 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.
+2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-01 Jan Schmidt <thaytan@mad.scientist.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.
- * configure.ac:
- Update to 0.10.11.2 (0.10.12 pre-release)
+2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Rewrote gen_tunnel_reply().
+ Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
+ than a hard coded string.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_async_play):
- base time is irrelevant here.
+2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * 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/audio/gstaudiosink.c: (audioringbuffer_thread_func):
- * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
- Improve debugging.
+2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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/rtsp/gstrtspconnection.c:
+ rtsp: Normalize lines (remove extra whitespace) before parsing.
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full):
- When we are asked to render N sample to 0 bytes, return N.
+2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-03-01 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made parse_string() return a result.
+ This will catch parsing errors when a too long string is received.
- * ext/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.
+2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * 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.
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
- * configure.ac:
- Convert to new AG_GST style.
+ * 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
-2007-02-28 Wim Taymans <wim@fluendo.com>
+2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- Patch by: Ed Catmur <ed at catmur dot co dot uk>
+ * 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/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.
+2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-02-28 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.
- * tests/check/Makefile.am:
- Include local stuff before system installed things in LDFLAGS and
- CFLAGS.
+2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-28 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/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
- Improve debugging.
+2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: fix includes for appsrc/appsink
- * 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.
+2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2007-02-28 Wim Taymans <wim@fluendo.com>
+ * 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).
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
- (gst_audio_clock_new):
- Fix clock name.
+2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_init), (gst_base_audio_sink_query):
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
- (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
- (gst_base_audio_src_create):
- Improve latency query code.
- Use proper clock names.
+ * ext/ogg/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.
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-08-18 21:45:31 +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/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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-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).
-2007-02-28 Stefan Kost <ensonic@users.sf.net>
+2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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-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...
-2007-02-27 Julien MOUTTE <julien@moutte.net>
+2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
- (msg_segment_done): Fix various seeking bugs (Slider was not
- updating when doing a non flushing seek, Reverse playback
- on segment seek was wrong).
+ * gst/playback/gstplaysink.c:
+ playsink: Also send SEEK events directly to a subpicture sink
-2007-02-26 Wim Taymans <wim@fluendo.com>
+2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (stop_seek):
- When we stop scrubbing, don't leave the pipeline PLAYING when we
- requested a PAUSED state.
+ * 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.
-2007-02-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: René Stadler <mail at renestadler de>
+ * 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-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Parse date strings in vorbis comments that have an invalid (zero)
- month or day (#410396).
+2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/tag.c: (GST_START_TEST):
- Test case for the above.
+ * 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.
-2007-02-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Loïc Minier <lool+gnome at via ecp fr>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post a missing plugin message additional to the error message on unknown types
+ Fixes bug #591677.
- * configure.ac:
- * ext/alsa/Makefile.am:
- * gst/audiotestsrc/Makefile.am:
- Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
+2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ playbin2: fix error message string
+ Fixes #591577.
- * gst/playback/gstplaybin.c:
- Improve docs: point out that the application needs to assist playbin
- with buffering.
+2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-02-23 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/riff/riff-read.c:
+ riff: align API doc of gst_riff_parse_chunk with reality
- * 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).
+2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2007-02-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid assertion failure on empty/NULL caps
- * 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.
-
-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 Stefan Kost <ensonic@users.sf.net>
-
- * 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
+2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2007-02-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Also detect SVG by the <svg> starting tag
+ Not all SVG images have the DOCTYPE specified.
- * tests/examples/seek/seek.c: (do_seek):
- Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
+2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.18 function
+ g_checksum_reset() was added only in GLib 2.18, but we still require
+ only 2.16, so work around that if we only have 2.16. Fixes #591357.
- * gst/volume/gstvolume.c: (volume_process_int16),
- (volume_process_int16_clamp), (volume_set_caps):
- Unbreak volume, value remains gint.
+2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-21 Stefan Kost <ensonic@users.sf.net>
+ * tests/check/pipelines/streamheader.c:
+ streamheader: Fix caps leak in the vorbisenc unit test
- * 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.
+2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-20 Edward Hervey <edward@fluendo.com>
+ * tests/check/pipelines/streamheader.c:
+ checks: fix stream header unit test hanging in gst_task_cleanup_all()
+ Set pipelines to NULL state and unref when done.
- * gst/playback/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.
+2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-20 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/md5.c:
+ * gst-libs/gst/rtsp/md5.h:
+ rtsp: Use GLib's GChecksum instead of our own MD5 implementation
- * gst/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
+2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix doc blurb typo for gst_navigation_send_key_event
- * 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).
+2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/subparse/gstsubparse.c:
+ subparse: Allow . instead of , as millisecond delimiter in srt subtitles
+ Fixes bug #591207.
- * gst-libs/gst/audio/audio.c:
- Fix documentation.
+2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-16 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- Patch by: Yves Lefebvre <ivanohe abacom com>
+2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
- Don't leak caps. Fixes #408278.
+ * 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.
-2007-02-15 Stefan Kost <ensonic@users.sf.net>
+2009-08-08 15:54:41 +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)
+ * 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.
-2007-02-15 Wim Taymans <wim@fluendo.com>
+2009-08-08 15:54:02 +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/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
-2007-02-15 Wim Taymans <wim@fluendo.com>
+2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.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.
+ * docs/design/draft-va.txt:
+ docs: add draft for generic introduction of video acceleration APIs idea
- * 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.
+2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-15 Wim Taymans <wim@fluendo.com>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ Revert "theora: Convert theoradec to libtheora 1.0 API"
+ This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
+ Temporarily revert until we have a workaround for debian/ubuntu
+ packaging failure (see http://bugs.debian.org/528710).
- * 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.
+2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-13 Stefan Kost <ensonic@users.sf.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.
- * configure.ac:
- Remove 'tests/examples/xerror/Makefile' from output files again.
+2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-13 Stefan Kost <ensonic@users.sf.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.
- * configure.ac:
- * docs/plugins/Makefile.am:
- Also crossref against gst-plugins-base-libs.
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
-2007-02-12 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: Convert theoradec to libtheora 1.0 API
- * configure.ac:
- * docs/libs/Makefile.am:
- * docs/plugins/Makefile.am:
- Add crossreferences to glib/gobject/gstream docs.
+2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/audio.h:
- Source formatting.
+ * ext/pango/gsttextrender.c:
+ textrender: Fix blitting of text over the output buffer and cairo painting
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
- Add own debug category.
+2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-12 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextrender.c:
+ textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
- Patch by: René Stadler <mail at renestadler de>
+2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/tag/gstvorbistag.c:
- Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
- (#403597).
+ * tests/icles/test-colorkey.c:
+ colorkey-test: fix xsync error
-2007-02-10 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
- * 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>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
- * 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 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-02-10 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/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).
+2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * tests/check/elements/subparse.c:
- Add test for this.
+ * 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.
-2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@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/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: change default slave method
+ Set the default slave method to the much better skew slaving algortihm.
- * 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).
+2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: make buffer writable
+ Make the input buffer writable before changing its contents.
- * 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).
+2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-02-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix postscript typefinder probability
+ Two bytes for a rare format hardly warrants MAXIMUM typefinding
+ probability, POSSIBLE seems more appropriate.
- Based on patch by: Julien Puydt <julien.puydt at laposte net>
+2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
- (gst_alsa_find_device_name):
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
- Improve device-name detection a bit, especially in the case where
- the device is not actually open (#405020, #405024). Move common code
- into gstalsa.c instead of duplicating it.
+ * ext/pango/gsttextoverlay.c:
+ pango: Send queries from the srcpad directly to the video sinkpad
-2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
+ * gst/subparse/gstsubparse.c:
+ subparse: Implement POSITION query
-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>
+2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- Also mention that a conversion from double to float is suboptimal still.
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ subparse: Implement SEEKING query
-2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
- * 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.
+ * configure.ac:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
+ Require latest core for this.
+ Fixes bug #590430.
-2007-02-05 Andy Wingo <wingo@pobox.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_callback): Update basesink->offset so that we
- pull monotonically increasing offsets instead of, um, seeking back
- to 0 each time. Fixes alsasrc ! alsasink!
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ pango: Add support for xRGB and BGRx formats
-2007-02-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@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).
+ * 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.
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
- Add small test to make sure request pads are cleaned up properly
- even if oggmux never changes state out of NULL.
+ * ext/pango/gsttextoverlay.c:
+ pango: Re-add shading support which was dropped by a previous patch
-2007-02-04 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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:
+ * ext/pango/gsttextoverlay.c:
+ pango: Check if pangocairo supports vertical rendering and fix properties
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_base_init),
- (gst_audio_filter_template_class_init),
- (gst_audio_filter_template_init),
- (gst_audio_filter_template_set_property),
- (gst_audio_filter_template_get_property),
- (gst_audio_filter_template_setup),
- (gst_audio_filter_template_filter),
- (gst_audio_filter_template_filter_inplace), (plugin_init):
- Oops, forgot to commit fixed-up example.
+ * ext/pango/gsttextrender.c:
+ textrender: Use PROP_X instead of ARG_X consistently
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
- (gst_audio_filter_class_init), (gst_audio_filter_init),
- (gst_audio_filter_set_caps),
- (gst_audio_filter_class_add_pad_templates):
- * gst-libs/gst/audio/gstaudiofilter.h:
- Port GstAudioFilter to 0.10. This change technically breaks
- API and ABI (and thus also every library developer's heart),
- but seems justifiable on the grounds that the base class was
- completely unusable before (ie. would crash immediately when
- actually used). Fixes #403963 (and eventually also #403572).
- Also document all of this a bit.
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ pango: Some minor cleanup
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@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.
+ * configure.ac:
+ pango: Check for pangocairo instead of pangoft2
-2007-02-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
- * tests/check/libs/utils.c:
- (test_base_utils_install_plugins_do_callout):
- Don't hard-code temp directory for test helper; use GLib functions
- to write out file and do error checking etc.
+ * 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.
-2007-02-02 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.
+2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- Add --with-install-plugins-helper configure option so distros can specify
- the path of the helper script or program to call when plugin installation
- is requested (distros: please do any argument munging in this helper
- script instead of patching GStreamer to pass arguments differently
- to another program directly).
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add SVG typefinder
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Build and document new API.
+2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/libs/utils.c: (result_cb),
- (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
- (libgstbaseutils_suite):
- Some simple checks for the new API.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add postscript typefinder
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/audioconvert.c: (test_float_conversion):
- Add small test for 32bit float <=> 64bit float conversion (works
- only one way so far, 32=>64 produces structured noise).
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Use static caps again for MPEG4 typefinding
-2007-02-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
- * 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/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Implement better & more flexible MPEG4 typefinding
+ This detects more MPEG4 streams as MPEG4.
+ Fixes bug #556537.
-2007-02-02 Stefan Kost <ensonic@users.sf.net>
+2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/audioconvert.c: (float), (double),
- (audio_convert_get_func_index):
- * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
- (make_lossless_changes):
- Support for 64-bit float audio in audioconvert (#339837)
+ * gst-libs/gst/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.
-2007-02-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Holger Wansing <linux wansing-online de>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Restrict width/height to 2^15 - 1
+ Otherwise integer overflows will happen, resulting in segmentation faults.
+ Fixes bug #590243.
- * po/LINGUAS:
- * po/de.po:
- Add German translation (#352069).
+2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-02-01 Sebastian Dröge <slomo@circular-chaos.org>
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix indention of template header
- reviewed by: Wim Taymans <wim@fluendo.com>
+2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.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).
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clarify documentation about caps and linkage
+ Fixes bug #589095.
-2007-01-31 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
- * gst/playback/gstplaybin.c: (gen_vis_element):
- Add audioresample+audioconvert in front of the visualisation
- element, so that elements like libvisual 0.4 that don't support all
- samplerates can work.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix typefinding of SDP files
+ Fixes bug #589574.
- Fixes: #402505
+2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
-2007-01-30 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Take the output offsets from the input if possible
+ Fixes bug #588915.
- * 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.
+2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-30 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Make sure to allocate enough memory for the temporary buffer
+ and fix scaling of odd-height interlaced video.
- * gst/audioconvert/gstaudioconvert.c:
- Don't fail on 0 sized buffers. Fixes #396835.
+2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 David Schleef <ds@schleef.org>
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix interlaced scaling for I420
+ ...and some other minor mistakes in the previous change.
- * gst/typefind/gsttypefindfunctions.c:
- Detect BBCD as video/x-dirac, so we can play raw dirac
- streams.
+2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-29 Wim Taymans <wim@fluendo.com>
+ * 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.
- * tests/check/gst/.cvsignore:
- Do as buildbot says.
+2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-29 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ back to development -> 0.10.24.1
- * 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.
+2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-27 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.doap:
+ Add 0.10.24 release to the doap file
- * 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.
+=== release 0.10.24 ===
-2007-01-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
-2007-01-25 Wim Taymans <wim@fluendo.com>
+2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstplaybasebin.c: (remove_source):
- Don't try to disconnect a signal from a finalized object.
+ * 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
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@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.
+ * 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.
-2007-01-25 Edward Hervey <edward@fluendo.com>
+2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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).
+ * ChangeLog:
+ * configure.ac:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.24.5 pre-release
-2007-01-25 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: (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/audio/gstaudiofilter.c:
+ audiofilter: Don't assert on slightly different caps
+ Plugins should not assert on incompatible caps, caps negotiation will
+ fail anyway.
-2007-01-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Make getting the current frame thread-safe.
+ * gst/adder/gstadder.c:
+ adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
-2007-01-25 Edward Hervey <edward@fluendo.com>
+2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@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.
+ * 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.
-2007-01-24 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
- * 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/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.
- * tests/check/gst-plugins-base.supp:
- Add suppression for pango on edgy/x86 for textoverlay test.
+2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2007-01-24 Wim Taymans <wim@fluendo.com>
+ * 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-libs/gst/rtp/gstrtpbuffer.h:
- Add some more fixed payloads.
+2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-23 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/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
- * 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).
+2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-23 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/v4l/.gitignore:
+ ignores: Ignore v4l probing example binary
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
- Don't leak mutex.
+2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@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_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.
+ * 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.
-2007-01-23 Edward Hervey <edward@fluendo.com>
+2009-07-24 00:42:16 +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
+ * common:
+ Automatic update of common submodule
+ From fedaaee to 94f95e3
-2007-01-23 Julien MOUTTE <julien@moutte.net>
+2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
- * 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-plugins-base.spec.in:
+ Update spec file with latest changes
-2007-01-22 Edward Hervey <edward@fluendo.com>
+2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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>.
+ * 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
-2007-01-22 Julien MOUTTE <julien@moutte.net>
+2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Lock that X Call as well. Fixes #397673.
+ * gst-libs/gst/audio/gstaudiosink.c:
+ audiosink: Add stream-status messages
+ Fixes #587695
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audiosrc: Add stream-status messages
+ See #587695
- * tests/check/Makefile.am:
- * tests/check/gst/.cvsignore:
- * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
- (typefindfunctions_suite):
- Add unit test for the above.
+2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/adder/gstadder.c:
+ gstadder: Don't forget to free pending events on flush/dispose.
+ Fixes #588747
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- On second thought, use "depth" field rather than "bpp" field.
+2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/adder.c:
+ tests/adder: Add stream consistency checking. Fixes #588748
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Camtasia caps apparently need a bpp field (#398875).
+2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
- * 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).
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ adder: Collect incoming tag events and send them after newsegment. Fixes #588747
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
- Plug a few reference leaks.
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Check for empty tag strings. Fixes #588724
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
-2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2007-01-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstchannelmix.c:
- Remove compatibility cruft for prehistoric GLib versions.
+ * 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.
-2007-01-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
- * 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).
+ * tests/examples/v4l/Makefile.am:
+ examples: fix v4l probe example build
+ Fixes bug #588550.
-2007-01-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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).
+ * 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
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * po/LINGUAS:
+ * po/tr.po:
+ Add Turkish translations
-2007-01-16 Stefan Kost <ensonic@users.sf.net>
+2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Red and blue mask was swapped (spotted by Dan Williams).
+ * 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.
-2007-01-15 Stefan Kost <ensonic@users.sf.net>
+2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Use new beats-per-minute tag from core.
+ * 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.
-2007-01-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
- * po/POTFILES.in:
- Add new files with translatable strings, so they actually make it
- into the template file one day.
+ * 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.
-2007-01-12 Andy Wingo <wingo@pobox.com>
+2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c
- (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
- (gst_base_audio_sink_activate_pull): Remove the handwavey nego
- stuff, as the base class handles this now. Actually tell the ring
- buffer to start.
- (gst_base_audio_sink_callback): Cast the ring buffer correctly.
- How did this work before? Maybe I'm not as awesome a programmer as
- I think.
+ * 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.
- * gst-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
- of a pad function.
+2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggmux.c:
+ oggmux: add Kate caps to the list of accepted types
+ See #525743.
- * 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.
+2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-01-12 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://...
- * 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.
+2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
-2007-01-12 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: simplify docs
+ Make short-desc short - its used in the toc. Strip uneeded markup.
- * gst/typefind/gsttypefindfunctions.c:
- Fix comment.
+2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-11 Tim-Philipp Müller <tim at centricular dot net>
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstvideo.def:
+ win32: Fix exports
+ Remove methods from video base classes that have moved to -bad.
+ Add gst_netaddress_to_string
- * 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).
+2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-10 Wim Taymans <wim@fluendo.com>
+ * tests/examples/gio/.gitignore:
+ ignores: ignore the giosrc-mounting example binary
- * 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.
+2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for Siren codec and make two descriptions non-translatable
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/utils/Makefile.am:
- * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
- * gst-libs/gst/utils/base-utils.h:
- * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
- (find_format_info), (caps_are_rtp_caps),
- (gst_base_utils_get_source_description),
- (gst_base_utils_get_sink_description),
- (gst_base_utils_get_decoder_description),
- (gst_base_utils_get_encoder_description),
- (gst_base_utils_get_element_description),
- (gst_base_utils_add_codec_description_to_tag_list),
- (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
- * gst-libs/gst/utils/descriptions.h:
- * gst-libs/gst/utils/missing-plugins.c:
- (missing_structure_get_type), (copy_and_clean_caps),
- (gst_missing_uri_source_message_new),
- (gst_missing_uri_sink_message_new),
- (gst_missing_element_message_new),
- (gst_missing_decoder_message_new),
- (gst_missing_encoder_message_new),
- (missing_structure_get_string_detail),
- (missing_structure_get_caps_detail),
- (gst_missing_plugin_message_get_installer_detail),
- (gst_missing_plugin_message_get_description),
- (gst_is_missing_plugin_message):
- * gst-libs/gst/utils/missing-plugins.h:
- API: add new libgstbaseutils library with functions
- - to create and parse missing-plugins messages
- - that provide (translated) descriptions for caps/decoders/sources/etc.
- Closes #392393.
+ * common:
+ Automatic update of common submodule
+ From 5845b63 to fedaaee
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-base.pc.in:
- Add new lib.
+2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Generate docs for new lib and API.
+ * 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.
- * 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.
+2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * ext/ogg/Makefile.am:
- Dist gstoggdemux.h to fix 'make distcheck'.
+2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
+ * configure.ac:
* sys/v4l/Makefile.am:
- Fix 'make distcheck' even more.
+ * sys/v4l/gstv4lelement.c:
+ v4lsrc: optional support for device probing with gudev
+ Enumerate v4l devices using gudev if available.
+ Fixes bug #583640.
-2007-01-09 Wim Taymans <wim@fluendo.com>
+2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
- (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
- (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
- (gst_ogg_demux_perform_seek):
- * ext/ogg/gstoggdemux.h:
- Added docs.
- Add some more comments.
- Small cleanups.
+ * gst/adder/gstadder.c:
+ adder: add since tags to docs
-2007-01-09 Wim Taymans <wim@fluendo.com>
+2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/theora/theoradec.c:
- * ext/vorbis/vorbisdec.c:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_commit_full):
- * gst-libs/gst/audio/gstringbuffer.h:
- * gst-libs/gst/rtp/gstrtpbuffer.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- Small documentation updates/fixes
+ * 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.
-2007-01-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- Require core CVS HEAD for Andy's basesrc/sink API additions.
+ * 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.
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Günter Thelen <daedalus dot inc at gmx net>
+ * tests/examples/seek/seek.c:
+ seek: add a checkbox for progressive download
- * 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).
+2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Fix template construction
+ Fix the construction of the temporary filename construction as the application
+ name can be NULL and we don't want a separator between the prgname and the
+ template.
- * configure.ac:
- Check if localtime_r() is available.
+2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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.
-2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
- * 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.
+2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2007-01-07 Julien MOUTTE <julien@moutte.net>
+ * 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.
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.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):
- Fixes: #390076.
- Add an adaptor property to select a specific XV adaptor.
- * sys/xvimage/xvimagesink.h:
+ * 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.
-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.
+2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
-2007-01-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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.
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
- (gst_ogm_parse_change_state):
- Activate pads before adding them to the element.
+2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiosink.c:
+ gio: Also post a "not-mounted" message from giosink
- * tests/examples/seek/scrubby.c: (main):
- * tests/examples/seek/seek.c: (main):
- Call g_thread_init() first thing in main() (see #391278).
+2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/gio/giosrc-mounting.c:
+ gio: Remove workaround for playbin2 bug in the sample application
+ The playbin2 bug was #588078.
- * 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.
+2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2007-01-05 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.
- * tests/check/Makefile.am:
- Update to use GST_PLUGINS_BASE_CFLAGS as well.
+2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * 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
-2007-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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
+ gio: Remove the experimental status from the GIO plugin
+ Fixes bug #510417.
-2007-01-04 Julien MOUTTE <julien@moutte.net>
+2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@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.
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ gio: Add documentation for the new "not-mounted" and "file-exists" messages
-2007-01-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * ext/gio/gstgiobasesrc.c:
+ gio: Make sure that we have the correct stream position when starting
-2007-01-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- * docs/Makefile.am:
- * docs/design/Makefile.am:
- Dist design docs.
+ * 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.
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation
- typo. Fixes: #390063.
+ * 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.
-2006-12-27 Julien MOUTTE <julien@moutte.net>
+2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
- caps leak.
- * win32/common/config.h: Updated.
+ * tests/check/pipelines/gio.c:
+ gio: Try to reuse the pipeline with the same stream objects
-2006-12-22 Stefan Kost <ensonic@users.sf.net>
+2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * ext/gio/gstgiobasesink.c:
+ * ext/gio/gstgiobasesrc.c:
+ gio: Improve the error message if a stream is already closed before usage
-2006-12-22 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
- (img_get_alpha_info):
- Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
- other end of the word. Fixes: #387073.
+ * 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.
- Add some inconsequential branch hints in a couple of places.
+2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-12-21 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiosrc.c:
+ gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
- * 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)
+2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-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
+ * ext/gio/gstgiosink.c:
+ gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
-2006-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Forgot to register the extensions.
+ * 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.
-2006-12-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
- * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
- (plugin_init):
- Add typefinder for VIVO files (my christmas present to the 90s).
+ * gst/audioconvert/gstchannelmix.c:
+ audioconvert: Fix compilation when debugging is disabled
+ Fixes bug #587980.
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (type_found):
- Special-case the text/plain media type: we only want to recognise it
- as a 'raw' decoded media type if it comes from a demuxer or subtitle
- parser, but not if the entire stream is of text/plain type. If the
- entire stream is text/plain, we should just error out.
+ * 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
- 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.
+2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- 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).
+ * 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
- Same problem also needs to be fixed for playbin+decodebin2.
+2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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
-2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
- * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
- Refuse to change state to READY when we failed to create any of the
- required elements in our instance init function.
+ * 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.
-2006-12-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Small docs fixes/updates.
+ * 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-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.
+2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
-2006-12-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * tests/check/elements/adder.c:
+ adder: make test more robust
+ Add audioconverts to the live-seeking test to make it negotiate.
- * Makefile.am:
- * gst-plugins-base.doap:
- * gst-plugins-base.spec.in:
- add doap file
+2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-12-09 Tim-Philipp Müller <tim at centricular dot net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: use core performance log category
- Patch by: Jens Granseuer <jensgr at gmx net>
+2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
- * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_handle_frame_based_buffer),
- (gst_base_rtp_audio_payload_handle_sample_based_buffer):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
- Declare variables at the beginning of a block. Fixes #383195.
+ * gst/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.
-2006-12-07 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
- * configure.ac:
- Bump version nano - back to CVS.
+ * ext/pango/gstclockoverlay.c:
+ pango: Call tzset() before localtime_r()
+ POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
+ required to set the state variables that define the current timezone. Indeed,
+ glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
+ if the system timezone is changed for a running program between two calls to
+ gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
+ timezone equals /etc/localtime being modified.
+ Fixes bug #587676.
+2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
-=== release 0.10.11 ===
+ * ext/Makefile.am:
+ build: remove spurious schroedinger reference
-2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
+2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
* configure.ac:
- releasing 0.10.11, "Dumb things"
+ * 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>
-2006-12-05 Jan Schmidt <thaytan@mad.scientist.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ netaddress: add constant for max len
- * 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.
+2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-04 Wim Taymans <wim@fluendo.com>
+ * 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()
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix #382223, add more dynamic caps handling.
+2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-12-04 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: make fd:// uri use buffering too
+ fd:// usually operate in push mode only and are thus suitable for buffering.
- * 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:
+2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-12-01 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ * gst/volume/gstvolume.c:
+ volume: include "1.0=100%" in property description
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Delete bad debug code.
- Fixes #381219
+2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-12-01 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/playback/gstplaysink.c:
+ playsink: remove unused property defs
- * 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
+2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-30 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/multichannel.c:
+ multichannel: rewrite the new doc comment a bit
+ Its part of the audio lib.
- * 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.
+2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-11-28 Wim Taymans <wim@fluendo.com>
+ * 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/playback/gstplaybasebin.c: (check_queue_event):
- Improve debug.
+2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-11-28 Wim Taymans <wim@fluendo.com>
+2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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'
-2006-11-28 Edward Hervey <edward@fluendo.com>
+2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: include in.h for FreeBSD compat
+ Fixes #586920
-2006-11-28 Wim Taymans <wim@fluendo.com>
+2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * win32/common/libgstapp.def:
+ defs: add defs for new appsink buffer-list method
-2006-11-26 Wim Taymans <wim@fluendo.com>
+2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
- * 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.
+2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2006-11-24 Wim Taymans <wim@fluendo.com>
+ * tests/check/elements/appsink.c:
+ Added unit tests for buffer list support in appsink.
- * 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.
+2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2006-11-24 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/app/gstappsink.c:
+ Added buffer list support.
- * 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.
+2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/app/gstappsink.h:
+ Added buffer list support.
- * gst/adder/gstadder.h:
- * gst/audiotestsrc/gstaudiotestsrc.h:
- remove obsolete _factory_init protos
+2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-11-24 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Include winsock2.h after defining WINVER.
+ Similar to bug #587080.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
- Fix spacing in debug message.
+2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-11-23 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Moved a comment.
- * 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.
+2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-23 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-22 Michael Smith <msmith@fluendo.com>
-
- * ext/theora/gsttheoraenc.h:
- * ext/theora/theoraenc.c: (gst_theora_enc_init),
- (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
- (theora_buffer_from_packet), (theora_enc_is_discontinuous),
- (theora_enc_chain), (theora_enc_change_state):
- Mark discontinuities of > 3/4 of a frame, reinit encoder.
+ * 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.
- * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
- (GST_START_TEST), (theoraenc_suite):
- Enable discontinuity test, fix it.
+2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
- * tests/check/Makefile.am:
- * tests/check/elements/.cvsignore:
- * tests/check/elements/textoverlay.c:
- (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
- (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
- (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
- (test_video_waits_for_text_send_text_newsegment_thread),
- (test_video_waits_for_text_shutdown_element),
- (test_render_continuity_push_video_buffers_thread),
- (textoverlay_suite):
- Add some unit tests for textoverlay.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Define WINVER before including any win headers
+ Fixes bug #587080.
-2006-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
- * 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.
+ * 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.
-2006-11-21 Stefan Kost <ensonic@users.sf.net>
+2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-sections.txt:
- Add some new symbols to the docs
+ * 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.
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-26 14:11:21 +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).
+ * 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
-2006-11-20 Wim Taymans <wim@fluendo.com>
+2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix RGBA32 caps. Fixes #357038.
+ * common:
+ Automatic update of common submodule
+ From f810030 to 5845b63
-2006-11-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/interfaces/mixertrack.h:
- Add FIXME so we can add some padding here in 0.11
+ * gst/playback/gstscreenshot.c:
+ screenshot: don't leak message
-2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
-2006-11-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From f3bb51b to f810030
- * 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.
+2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-11-16 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for multipart
+ So we get slightly nicer error messages when multipartdemux is missing.
- * 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.
+2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- Enable audiorate test now that it passes.
+ * 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.
-2006-11-09 Stefan Kost <ensonic@users.sf.net>
+2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- clear xv when going to NULL, remove // commented non-existant proto
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix leak when the source fails to change state
- * tests/examples/seek/seek.c: (main):
- add missing tooltip description for scrub and play_scrub
+2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-11-14 David Schleef <ds@schleef.org>
+ * gst/subparse/gstssaparse.c:
+ ssaparse: avoid leaking all buffers
- * 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.
+2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-14 David Schleef <ds@schleef.org>
+ * 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)
- * 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.
+2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-11-14 Wim Taymans <wim@fluendo.com>
+ * 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.
- Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
+2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2006-11-13 Wim Taymans <wim@fluendo.com>
+2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@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.
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
+ Also make all the function arrays constant.
-2006-11-13 Michael Smith <msmith@fluendo.com>
+2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
- Improve a debug line slightly.
+ * 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.
- * 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).
+2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-11-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix NV12 and NV21 transformations
+ Fix some stride problems, fix the nv12 to nv21 direct transformation,
+ and implement a direct conversion to yuv444 to save CPU.
- Patch by: Sergey Scobich <sergery.scobich at gmail com>
+2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
- * win32/common/libgsttag.def:
- Add missing symbol (#366492).
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix NV12 painting for odd strides/heights
-2006-11-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
- Don't unref a NULL pad.
+ * 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.
-2006-11-09 Wim Taymans <wim@fluendo.com>
+2009-06-19 21:25:54 +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.
+ * 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.
-2006-11-07 Stefan Kost <ensonic@users.sf.net>
+2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
- * 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
+ * configure.ac:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ cdparanoia: expose cache size setting
+ This setting was added in cdparanoia 10.2. The default value is good
+ for audio extraction, but lower values (previous versions of cdparanoia
+ used 150) are better for realtime playback.
+ Fixes #586331.
-2006-11-06 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@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-plugins-base.spec.in:
+ Make build of schro plugin conditional
-2006-11-03 David Schleef <ds@schleef.org>
+2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggmux.c:
- Add/remove KW-DIRAC header here, since it is ogg-specific.
+ * 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
-2006-11-03 Michael Smith <msmith@fluendo.com>
+2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
- Recognise more mpeg4 elementary video streams.
+ * 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...
-2006-11-02 Edward Hervey <edward@fluendo.com>
+2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * win32/common/libgstrtp.def:
+ defs: add new symbol to win32 defs file
+ Based on patches by Ognyan Tonchev.
+ See #585559
-2006-11-02 Wim Taymans <wim@fluendo.com>
+2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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
-2006-11-02 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Sergey Scobich <sergey.scobich at gmail 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
- * 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).
+2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-11-01 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtp: add bufferlist support
- * ext/ogg/gstogg.c:
- Remove unused variable.
+2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c:
- Fix Wim's surname in plugin description.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ rtp: pass data to macros instead of GstBuffer
-2006-10-31 Wim Taymans <wim@fluendo.com>
+2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-plugins-base.spec.in:
- spec new .h file. Fixes #368310.
+ * 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
-2006-10-31 Michael Smith <msmith@fluendo.com>
+2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: appname might be NULL
+ Don't set title if appname is unknown.
-2006-10-31 Wim Taymans <wim@fluendo.com>
+2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/rtp/Makefile.am:
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- (gst_base_rtp_audio_payload_init):
- Fix and activate base audio payloader.
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: set window title from application name
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
- (plugin_init):
- Add typefinder for QuickTime Image Files (see #366156).
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Made the parsing of the RTSP URL scheme more generic.
-2006-10-28 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
- Another typo fix (#366212).
+ * 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()
-2006-10-27 Wim Taymans <wim@fluendo.com>
+2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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/rtsp/gstrtspconnection.c:
+ rtsp: Only extract the session ID from RTSP responses.
-2006-10-27 Wim Taymans <wim@fluendo.com>
+2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
- Patch by: Mark Nauwelaerts <manauw at skynet dot be>
+ * gst-libs/gst/rtsp/gstrtspurl.c:
+ rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
- Remove the pad from the element in release_pad. Fixes #364812.
+2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
- * 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).
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-26 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-23 Tim-Philipp Müller <tim at centricular dot 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/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.
+2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/subparse.c: (test_tmplayer_do_test),
- (GST_START_TEST), (subparse_suite):
- Add some basic unit tests for the above.
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ audio: correctly handle short read/writes
-2006-10-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
- * 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.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: add some extra logging for buffer timestamps
-2006-10-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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).
+ * 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>
-2006-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Free iterator after removing all groups
- * 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.).
+2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/subparse.c: (test_srt_do_test),
- (test_srt):
- Some more tests for .srt incl. tests for the above stuff.
+ * gst-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.
-2006-10-20 Julien MOUTTE <julien@moutte.net>
+2009-06-16 18:57:20 +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-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add Timestamp header field
+ fixes #585994
-2006-10-20 Michael Smith <msmith@fluendo.com>
+2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@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:
+ 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
-2006-10-20 Michael Smith <msmith@fluendo.com>
+2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/playback/gstplaysink.c:
+ playsink: set the sink flag on the element
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
- (plugin_init):
- Typefind mmsh header data packet to application/x-mmsh (#362625).
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: add debug message
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-10-19 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
- (parse_subrip):
- Strip trailing newlines from subtitle text output.
+ * 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).
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
- (gst_sub_parse_change_state):
- Fix memleak; clear subparse->textbuf n state change function.
+ * 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.
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't require subrip (.srt) files to start with a chunk number of 1.
+ * tests/examples/seek/seek.c:
+ Don't use deprecated GTK API
+ Fixes bug #585758.
-2006-10-18 Wim Taymans <wim@fluendo.com>
+2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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()
+ * 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.
- * 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.
+2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use a more consistent naming of GstRTSPRec variables.
- * 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.
+2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
- * tests/check/elements/playbin.c: (playbin_suite):
- Enable unit test for the above.
+ * 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.
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/playback/gstplaybin.c: (setup_sinks):
- The old pad activation spiel.
+ * 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.
-2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (setup_source):
- Don't hang forever if the subbin already fails to start up in
- the state change to PAUSED (#339366).
+ * 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)
-2006-10-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
- (gst_tuner_set_channel), (gst_tuner_get_channel),
- (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
- (gst_tuner_set_frequency), (gst_tuner_get_frequency),
- (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
- (gst_tuner_find_channel_by_name):
- Fix some function guards, add some more function guards.
+ * tests/examples/seek/seek.c:
+ seek: add shuttle controls
-2006-10-17 Jan Schmidt <thaytan@mad.scientist.com>
+2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
- (remove_element_chain):
- Don't return a pad from get_our_ghost_pad unless it is actually the
- one we want.
- Change a cast in remove_element_chain slightly.
+ * tests/examples/seek/stepping2.c:
+ example: fix compile
-2006-10-13 Julien MOUTTE <julien@moutte.net>
+2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek), (start_seek),
- (rate_spinbutton_changed_cb), (segment_done), (msg_state_changed):
- Segment seeking needs to use the rate and set stop to -1.
+ * tests/examples/seek/Makefile.am:
+ examples: build the stepping2 example
-2006-10-13 Wim Taymans <wim@fluendo.com>
+2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_setcaps):
- Don't crash when ringbuffer is not yet created.
- Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
- Fixes #361634.
+ * gst/playback/gstplaysink.c:
+ playsink: update for new step API
- * 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.
+2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-13 Julien MOUTTE <julien@moutte.net>
+ * 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.
- * 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.
+2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-13 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-12 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: handle border cases in resampler
- * 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 !
+2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-10-12 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ docs: Update common. Use upload-doc.mak instead of upload.mak
- Patch by: Sebastien Cote <sebas642 at yahoo.ca>
+2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_finalize):
- Fix two small memory leaks (#361456).
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ docs: fix typo
-2006-10-10 Julien MOUTTE <julien@moutte.net>
+2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/examples/seek/seek.c: (do_seek),
- (rate_spinbutton_changed_cb): When changing spinbutton we try
- to change the rate on the fly.
+ * gst-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.
-2006-10-10 Wim Taymans <wim@fluendo.com>
+2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.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):
- Add WMS caps.
+ * 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.
-2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Josep Torra Valles <josep@fluendo.com>
+ * gst-libs/gst/audio/testchannels.c:
+ Don't include config.h multiple times when build audio testchannel app.
+ Fixes build problem on win32 (#585075).
- * 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.
+2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-10-10 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gsturidecodebin.c:
+ playbin2/uridecodebin: Fix connection-speed propagation
+ uridecodebin expects the passed connection-speed value in kbps, so we
+ need to divide the value stored in bps by 1000. Also, lower the upper
+ limit on the properties to the value that we can actually store in our
+ internal guint (which is plenty high enough)
- Patch by: Ferenc Gerlits <fgerlits at gmail com>
+2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c:
- Recognise XML files and XML-like files shorter than 256 bytes as
- well (fixes #359237).
+ * 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.
-2006-10-09 Edgard Lima <edgard.lima@indt.org.br>
+2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- Patch by: Renato Filho <renato.filho@indt.org.br>
-
- * gst/typefind/gsttypefindfunctions.c:
- Added typefind functions to video/x-nuv media.
-
-2006-10-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtsptransport.h:
+ rtsp: add some more docs
- * gst-libs/gst/interfaces/xoverlay.c:
- (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
- Some more guards against invalid input.
+2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-07 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: Avoid a compiler warning.
- * 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 !)
+2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-06 Stefan Kost <ensonic@users.sf.net>
+ * 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-libs/gst/interfaces/xoverlay.c:
- Unref message in doc-example (spotted by Robert McQueen)
+2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ autogen: remove -Wno-portability from here
+ as it is in configure.ac now.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- printf fix.
+2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Plug a memory leak.
+ Free memory related to any partially read and/or written RTSP messages.
- * 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.
+2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-06 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: no need to cause discont when clipping
+ Remove the discont-when-clipping hack now that basesink provides us with
+ correctly clipped samples when stepping.
- * gst-libs/gst/floatcast/floatcast.h:
- Fix obviously-bogus macros; use the correct types.
+2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-10-06 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_change_state):
- Also call parent state change function to activate pads.
+2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (mpeg1_parse_header), (mpeg1_sys_type_find):
- Add some more debug info in mpeg typefinding.
+ * 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.
-2006-10-06 Michael Smith <msmith@fluendo.com>
+2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
- * ext/theora/theoradec.c: (theora_dec_chain):
- Zero byte theora packets are valid and well-defined; don't warn on
- them.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add description for hdv/aux-* formats.
-2006-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
- * 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
+ * ext/schroedinger/Makefile.am:
+ Added libgstbase to schro's LIBADD
+ Fixes #585079
-2006-10-05 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/alsa/gstalsadeviceprobe.c:
- (gst_alsa_device_property_probe_get_values):
- * ext/alsa/gstalsasink.c: (set_hwparams):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
- (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
- (gst_ogg_mux_process_best_pad):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
- (gst_ogg_parse_chain):
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
- (gst_vorbis_enc_buffer_check_discontinuous):
- * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_handle_track_seek):
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_push_full):
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- * gst/audioresample/resample.c: (resample_input_pushthrough):
- * gst/playback/gstplaybasebin.c: (queue_out_of_data):
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
- (wavpack_type_find):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
- (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
- * tests/check/elements/volume.c: (GST_START_TEST):
- Printf format fixes.
+ * gst-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.
-2006-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
- Fix a simple mistake (see the docs)
- Fixes #359580
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: make sure varargs are of right type
+ Explicitly cast the variables to g_object_set to their right types.
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-05 16:49:58 +0200 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.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.
+ * 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.
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
- Patch by: James "Doc" Livingston <doclivingston at gmail com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Fixed a typo.
- * 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).
+2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Remove an unused variable.
-2006-10-03 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Removed duplicate initialization of conn->writefd.
-2006-09-30 Jan Schmidt <thaytan@mad.scientist.com>
+2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Fix typo in a debug statement.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Use #defined status codes.
- * 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.
+2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
- Move a recurring message to LOG level.
+ * 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.
- * 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.
+2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * configure.ac:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
+ See #584835. Also update win32 files while we're at it.
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- Removed empty * between paragraphs
+2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst/playback/gstplaybin2.c:
+ playbin2: API: Add {audio,video,text}-tags-changed signals
+ Fixes bug #584686.
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- * gst-libs/gst/rtp/README:
- Moved some documentation into .c file
+2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-29 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/playback/gstdecodebin.c: (no_more_pads):
- Fix compilation.
+2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
+ * tests/examples/seek/.gitignore:
+ ignore: add new stepping app to ignore
- * gst/playback/gstdecodebin.c: (new_caps):
- Remove g_print
+2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Add some docs.
+ * 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.
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/Makefile.am:
- Re-enable cddabasesrc test to see if it works again
- now.
+ * 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.
-2006-09-29 Wim Taymans <wim@fluendo.com>
+2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (setup_subtitle),
- (gen_source_element):
- Handle invalid URIs a bit more gracefully.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: add G_LIKELY because we can
-2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
- * tests/check/pipelines/oggmux.c:
- Remove obsolete comment.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix caps for ogg typefinder.
-2006-09-29 Michael Smith <msmith@fluendo.com>
+2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
- (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
- (gst_ogg_mux_collected):
- Commit patch from James "Doc" Livingston, adds proper EOS handling
- in oggmux. GStreamer can, for the first time ever, create a valid
- Ogg file! Yay!
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: remove some cruft from -sections.txt file
- * tests/check/pipelines/oggmux.c: (check_chain_final_state),
- (oggmux_suite):
- Reenable tests now that they pass.
+2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-29 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
- Stop reading commands when EOF (we read 0) as well.
-
-2006-09-28 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
- (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
- (find_dynamic), (unlinked), (close_link):
- Implement delayed caps linking needed for element with a lot of
- different caps on the src pads that get fixed at runtime.
- Improve management of dynamic elements.
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (group_destroy), (group_commit), (check_queue), (queue_overrun),
- (gen_preroll_element), (remove_groups), (unknown_type),
- (add_element_stream), (no_more_pads_full), (no_more_pads),
- (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
- (new_decoded_pad), (setup_subtitle), (array_has_value),
- (gen_source_element), (source_new_pad), (has_all_raw_caps),
- (analyse_source), (remove_decoders), (make_decoder),
- (remove_source), (setup_source), (finish_source), (prepare_output),
- (gst_play_base_bin_change_state):
- * gst/playback/gstplaybasebin.h:
- Use more _CAST instead of full type checking casts.
- Small cleanups, plug some leaks.
- Handle dynamic sources.
- Add some helper functions to create lists of strings used for
- blacklisting and other stuff.
- Refactor some code dealing with analysing the source.
- Re-enable sources without pads (like cd:// or other selfcontained
- elements).
+ * gst/playback/gstplaysink.c:
+ * tests/examples/seek/seek.c:
+ add framestepping to playbin2 and seek
-2006-09-28 Wim Taymans <wim@fluendo.com>
+2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
- * 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/rtsp/gstrtspconnection.c:
+ rtsp: Avoid compiler warnings with -Wextra.
-2006-09-28 Wim Taymans <wim@fluendo.com>
+2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
- Set caps on outgoing buffers.
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
- * 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
+2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
-2006-09-28 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ sdp: Remove an unused variable.
- * 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?)
+2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-28 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
- * 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.
+2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-09-28 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.
- * 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).
+2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-09-27 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Add some more info in a WARNING.
+2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- Handle PAUSE in create function, use new -core addition to
- wait for playing. Fixes pausing and resuming capture from an
- audiosrc.
+ * gst/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-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Constify some more.
- Caller supports interrupted reads now.
+2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstplaysink.c:
+ playbin2: cosmetic change to avoid unnecessary line breaks
+ Looks nicer and works around gst-indent silliness.
- * tests/check/Makefile.am:
- Another attempt to make the gen64 buildbot happy.
+2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-27 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
+2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_clear_actors),
- (gst_visual_chain), (gst_visual_change_state):
- Libvisual plugin was not passing audio data to libvisual 0.4.0
- correctly. Fixes #357800
+ * gst/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>
-2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/examples/seek/seek.c:
+ seek: add volume label and sync with sink volume
+ Look at the volume and have the pulsemixer open at same time. Unfortunately
+ playbin2 does not emit notify on volume right, so this polls for now.
- * 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.
+2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-27 Wim Taymans <wim@fluendo.com>
+ * 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-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.
+2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-27 Wim Taymans <wim@fluendo.com>
+ * 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
- * 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.
+2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-27 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
- * tests/check/Makefile.am:
- Temporarily disable test that fails on the bots for unknown reasons.
+2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-26 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ netbuffer: also note the order of IP4 addresses
+ IP4 addresses are also stored in network byte order. Make a note of this in the
+ docs.
- * 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>
+2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
- * ChangeLog:
- ChangeLog surgery for missing bug-number
+ * ext/theora/theoraparse.c:
+ theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- Patch by: Peter Kjellerstedt <pkj at axis 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/playback/test.c:
- Fix compilation with uClibc and -Werror (#357591).
+2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
-2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
+ * common:
+ Update common
- * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
- Parse dates that are followed by a time as well (#357532).
+2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/libs/tag.c: (test_vorbis_tags):
- Add unit test for this.
+ * 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.
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
- (gst_audio_convert_transform_caps):
- * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
- * gst/videotestsrc/videotestsrc.h:
- A few array const-ifications.
+ * gst/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
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/Makefile.am:
- See if this makes the build bots happy.
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
- * tests/check/libs/cddabasesrc.c:
- UTF8-ise my name.
+2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-23 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
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.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).
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/Makefile.am:
- Need to build tag directory before cdda.
+ * 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.
-2006-09-23 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.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/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
- * gst-libs/gst/tag/gstvorbistag.c:
- Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
- tags (also see #347848).
+2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
- Log vorbis comments we are actually writing. Const-ify array.
+ * gst-plugins-base.spec.in:
+ Update spec file
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (gen_preroll_element):
- Improve buffering a bit by avoiding a deadlock because we cannot assume
- the underrun is always called.
+ * 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.
-2006-09-23 Wim Taymans <wim@fluendo.com>
+2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
- Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ basevideo: Fix memleak
- * 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.
+2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
-2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/schroedinger/gstschrodec.c:
+ * ext/schroedinger/gstschroparse.c:
+ schro: Fix usage of adapter_masked_scan_uint32
+ Because *somebody* changed the API without telling me.
- * ext/gnomevfs/gstgnomevfssrc.c:
- Fix misleading docs addition.
+2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
- * tests/check/elements/videotestsrc.c: (check_rgb_buf):
- Get rid of compiler warning the right way.
+ * ext/schroedinger/gstschro.c:
+ schro: Change package name to GST_PACKAGE_NAME
-2006-09-22 Wim Taymans <wim@fluendo.com>
+2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
- * 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/video/gstbasevideoencoder.c:
+ basevideo: Add preset interface to encoder
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
- timestamps are uint.
+2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
-2006-09-22 Stefan Kost <ensonic@users.sf.net>
+ * gst/audioresample/gstaudioresample.c:
+ Run liboil benchmark multiple times
+ The statistics function requires multiple runs, otherwise
+ it causes a divide by zero error.
- * gst-libs/gst/interfaces/xoverlay.c:
- Remove unused statement from doc example.
+2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-21 Stefan Kost <ensonic@users.sf.net>
-
- * gst-libs/gst/interfaces/videoorientation.c:
- (gst_video_orientation_iface_init),
- (gst_video_orientation_get_hflip),
- (gst_video_orientation_get_vflip),
- (gst_video_orientation_get_hcenter),
- (gst_video_orientation_get_vcenter),
- (gst_video_orientation_set_hflip),
- (gst_video_orientation_set_vflip),
- (gst_video_orientation_set_hcenter),
- (gst_video_orientation_set_vcenter):
- Add since tags to new API docs, ChangeLog surgery (forgot API keyword
- in ChangeLog)
+ * 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).
-2006-09-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-22 17:41:50 +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).
+ * 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.
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybasebin.c: (group_commit),
- (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
- (queue_out_of_data), (gen_preroll_element),
- (preroll_remove_overrun), (probe_triggered):
- Refactor handling of overrun detection.
- Separate handling of group completion and deadlock detection when doing
- network buffering. This should fix some deadlocks that were not detected
- because the group was completed.
- Add more comments, improve debugging.
+ * gst/playback/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.
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/libs/audio.c:
- Some more compilation fixes.
+ * common:
+ Automatic update of common submodule
+ From d3a8fab to 888e0a2
-2006-09-21 Wim Taymans <wim@fluendo.com>
+2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Early morning compilation fix.
+ * win32/common/libgstvideo.def:
+ win32: Remove gst_adapter_masked_scan_uint32 from the exports
-2006-09-20 Wim Taymans <wim@fluendo.com>
+2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * tests/check/elements/gdpdepay.c: (GST_START_TEST):
- * tests/check/elements/multifdsink.c: (GST_START_TEST):
- * tests/check/elements/videorate.c: (GST_START_TEST):
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
- Fix some warnings.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ audiosink: improve debug message
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
- * 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.
+ * 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.
-2006-09-20 Stefan Kost <ensonic@users.sf.net>
+2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_get_times):
- change colorkey behaviour back according to #354773 comment 6/7
+ * 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.
-2006-09-19 Michael Smith <msmith@fluendo.com>
+2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
- (gst_multi_fd_sink_recover_client),
- (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
- (gst_multi_fd_sink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Implement stubbed out properties unit-type, units-soft-max,
- units-max, to allow specifying maximum sizes in units other than
- buffers.
- Fixes #355935
+ * win32/common/libgstvideo.def:
+ defs: add new symbols
-2006-09-19 Wim Taymans <wim@fluendo.com>
+2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- Reorder the audio formats a bit for clarity.
- Detect and create caps for MSGSM and MSN (WAV49).
- Fixes #356596.
+ * configure.ac:
+ autogen: pass -Wno-portability to automake to suppress warnings
+ GNU make is needed.
- * 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.
+2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * docs/libs/.gitignore:
+ gitignore: remove bogus *.sgml wildcard - these files are tracked in git
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs.types:
- * gst-libs/gst/interfaces/Makefile.am:
- * gst-libs/gst/interfaces/videoorientation.c:
- (gst_video_orientation_get_type),
- (gst_video_orientation_iface_init),
- (gst_video_orientation_get_hflip),
- (gst_video_orientation_get_vflip),
- (gst_video_orientation_get_hcenter),
- (gst_video_orientation_get_vcenter),
- (gst_video_orientation_set_hflip),
- (gst_video_orientation_set_vflip),
- (gst_video_orientation_set_hcenter),
- (gst_video_orientation_set_vcenter):
- * gst-libs/gst/interfaces/videoorientation.h:
- API: Add new interface to control video orientation (fixes #354908)
+2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * gst/tcp/gsttcpclientsrc.c:
+ tcpclientsrc: this is not a live source
+ Don't mark us as a live source because we are not.
- * 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().
+2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * gst/adder/gstadder.c:
+ adder: only send flush_stop when seek failed
+ This is still not the ultimate fix. Added some comment to explain the troubles.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
- Give better warning message (add object and detail).
+2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * 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()
+2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-18 Michael Smith <msmith@fluendo.com>
+ * 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/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.
+2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-18 Stefan Kost <ensonic@users.sf.net>
+ * 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).
- * 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
+2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
- * tests/examples/seek/seek.c: (main):
- add real meadi-buttons, add tool-tips for the seek-options, arrange
- seek options in a table
+ * gst/playback/gstdecodebin.c:
+ decodebin: use iterators instead of list
+ The list api is deprecated. Use threadsafe iterators instead.
-2006-09-18 Michael Smith <msmith@fluendo.com>
+2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
- (gst_ogg_mux_push_buffer):
- Don't generate out-of-order timestamps from oggmux, instead clamp
- output timestamps to be >= the previously output ts.
- Fixes #355595
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: configure caps on decodebin2
+ Implement the caps property by setting the configured caps on new decodebin2
+ objects.
+ Fixes #582749
-2006-09-18 Michael Smith <msmith@fluendo.com>
+2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multi_fd_sink_class_init):
- Updates, fixes, and typo corrections for multifdsink. No functional
- changes.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid some _caps_ref in some cases
+ Only mess with the caps refcount when we configure different caps.
-2006-09-17 Michael Smith <msmith@fluendo.com>
+2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
- Don't crash on truncated files - check that we got an 8 byte buffer
- before trying to memcmp it.
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: fix potential caps leak
+ Free the user-configured caps in finalize.
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (get_active_source):
- Make stream-switching appear instant to the application
- (ie. make sure that a g_object_get on 'current-foo' returns
- the stream previously set with g_object_set(). Totem needs
- this to update stream-related meta-info (like audio-codec)
- correctly when switching streams.
+ * gst/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
-2006-09-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
- (gst_alsa_mixer_ensure_track_list):
- Try harder to guess which mixer track is the master mixer
- track (instead of just taking the first one that has a pvolume).
- Fixes #342228.
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: 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
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
- reviewed by: <delete if not using a buddy>
+ * 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.
- * gst-libs/gst/audio/audio.h:
- * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
+2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * ext/theora/theoradec.c:
+ theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
- * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
- (gst_audio_convert_transform_caps):
- Get structure-name just once.
+2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * gst/adder/gstadder.c:
+ adder: add more logging and return value checking
- * 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.
+2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-17 Stefan Kost <ensonic@users.sf.net>
+ * gst/adder/gstadder.c:
+ adder: handle the return value from iterator_fold
- * ext/gnomevfs/gstgnomevfssrc.c:
- Add docs about icydemux usage in connection with gnomevfssrc
+2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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/adder/gstadder.c:
+ adder: use the pad in logging as objects
+ Helps to differenciate between source and sinks pads.
- * gst/audiotestsrc/gstaudiotestsrc.h:
- Fix wrong info in header due to copy & paste
+2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
-2006-09-15 Wim Taymans <wim@fluendo.com>
+ * tests/examples/seek/seek.c:
+ seek: use parser for mp3 and rename variable
- * 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.
+2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-15 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.
- * configure.ac:
- We require 0.10.10.1 now because of _wait_preroll().
+2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Use gst_base_sink_wait_preroll().
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for v210 and v216 formats
-2006-09-15 Wim Taymans <wim@fluendo.com>
+2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
- * 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/video/gstbasevideocodec.c:
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ video: remove // comments
-=== release 0.10.10 ===
+2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add Y444, v210, v216 formats
- patch by: Michael Smith <msmith at fluendo dot com>
+2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
- * 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
+ * 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>
-2006-09-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/gstbasevideocodec.c:
+ * gst-libs/gst/video/gstbasevideocodec.h:
+ * gst-libs/gst/video/gstbasevideodecoder.c:
+ * gst-libs/gst/video/gstbasevideodecoder.h:
+ * gst-libs/gst/video/gstbasevideoencoder.c:
+ * gst-libs/gst/video/gstbasevideoencoder.h:
+ * gst-libs/gst/video/gstbasevideoparse.c:
+ * gst-libs/gst/video/gstbasevideoparse.h:
+ * gst-libs/gst/video/gstbasevideoutils.c:
+ * gst-libs/gst/video/gstbasevideoutils.h:
+ video: Copy BaseVideo classes from Schroedinger
+
+2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
- patch by: Wim Taymans <wim at fluendo dot com>
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: add num-fds property
+ multifdsink::num-fds
- * 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
+2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-05 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
- * tests/check/Makefile.am:
- Random variation in Makefile line to see if it makes the
- gen64-base-full bot any happier.
+2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Implement Preset interface
- * tests/check/pipelines/oggmux.c: (oggmux_suite):
- Disable test that fails at the moment (killed after timeout).
+2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/theoraenc.c:
+ theoraenc: Implement Preset interface
- Patch by: James Livingston <doclivingston at gmail.com>
+2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/ogg/gstoggmux.c:
+ oggmux: Implement Preset interface
-2006-09-04 Wim Taymans <wim@fluendo.com>
+2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
- patch by: Alessandro Dessina <alessandro nnva org>
+ * gst/playback/gstplaysink.c:
+ playbin2: Fix cdda:// playback
+ Don't send async-start when the playsink has already been configured
+ before changing state.
- * ext/ogg/gstoggmux.c:
- Add cmml caps to oggmux. Fixes #353912
+2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+ * configure.ac:
+ configure: require core CVS for gst_adapter_prev_timestamp()
+ which is used in the libvisual plugin.
- * 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.
+2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-09-02 Tim-Philipp Müller <tim at centricular dot net>
+ * AUTHORS:
+ AUTHORS: fix my email
- * 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.
+2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: make our internal time monotonic
+ Make the internal time increase monotonically.
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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").
+ * ext/libvisual/visual.c:
+ visual: remove next_ts variable
+ We can remove the next_ts variable as we don't use it anymore.
-2006-09-01 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
- Fix stride for YVYU, should be word-aligned (#353658).
+ * 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.
-2006-08-31 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/adder/gstadder.c: (gst_adder_src_event):
- Fix build.
+ * 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.
-2006-08-31 Edward Hervey <edward@fluendo.com>
+2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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).
+ * 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.
-2006-08-31 Edward Hervey <edward@fluendo.com>
+2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@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/audiotestsrc/gstaudiotestsrc.c:
+ audiotestsrc: seek to the requested byte offset, not the expected byte offset
-2006-08-30 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@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/audiotestsrc/gstaudiotestsrc.c:
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ audiotestsrc: support more than just one channel
-2006-08-29 Wim Taymans <wim@fluendo.com>
+2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ propertyprobe: Fix typo in the docs
- * 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.
+2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
-2006-08-29 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theora.c:
+ * ext/vorbis/vorbis.c:
+ Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
- Patch by: Viktor Peters <viktor dot peters at gmail dot com>
+2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2006-08-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybin.c:
- Improve docs: add section about BUFFERING messages sent by playbin.
+ * ext/ogg/gstoggmux.c:
+ oggmux: small debug statement in DISCONT
-2006-08-29 Michael Smith <msmith@fluendo.com>
+2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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>
-2006-08-29 Edward Hervey <edward@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/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.
+2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ rtpdepay: add some more comments
- * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
- fix buffer unreffing on a header push failure
+2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: make sure values are ever increasing
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
- (gst_audio_rate_chain):
- Make the metadata of the buffer writable before changing its
- flags.
+2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-28 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-25 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_multi_fd_sink_render):
- Fix spelling.
- Remove accidently included debug line.
+2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2006-08-25 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/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.
+2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-25 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/interfaces/tunernorm.h:
+ interfaces: Seperate some more struct definitions from typedefs
- * 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.
+2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
- (gst_multi_fd_sink_render):
- Do not use gst_buffer_make_writable() in a basesink render method,
- as it may incorrectly unref the buffer. Instead, use convoluted
- dance to avoid copying the buffer except when we need to.
+ * gst-libs/gst/interfaces/navigation.h:
+ * gst-libs/gst/interfaces/videoorientation.h:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ interfaces: Seperate some more struct definitions from typedefs
-2006-08-25 Michael Smith <msmith@fluendo.com>
+2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vorbis/vorbisenc.c:
- (gst_vorbis_enc_buffer_check_discontinuous):
- Allow very small discontinuities in the timestamps. These we can't
- do anything useful with anyway (because vorbis's timestamps have
- only sample granularity), and are commonly produced by elements with
- minor bugs. Allow up to 1/2 a sample out.
- Fixes #351742.
+ * win32/common/libgstinterfaces.def:
+ Add new functions to win32 exports
-2006-08-24 Wim Taymans <wim@fluendo.com>
+2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
- (play_scrub_toggle_cb), (main):
- Add a checkbox to enable play scrubbing. Makes it possible to disable
- normal scrubbing.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add new functions to the docs
-2006-08-23 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * tests/check/elements/.cvsignore:
- make buildbot happy
+ * 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.
-2006-08-23 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.
+2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-23 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/interfaces/colorbalance.c:
+ interfaces: Add docs for gst_color_balance_get_balance_type()
- * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
- Waits for tasks to settle down so that we clean up correctly for
- valgrind.
+2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+ * autogen.sh:
+ Run libtoolize before aclocal
+ This unbreaks the build in some cases. Fixes bug #582021
- * 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.
+2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/pango/gsttextrender.c:
+ textrender: Correctly initialize the background for ARGB too
- * 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.
+2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * 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.
-2006-08-22 Wim Taymans <wim@fluendo.com>
+2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@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.
+ * ext/pango/gsttextrender.c:
+ textrender: Don't always use ARGB if downstream supports it but take it's preference
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
- * 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).
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ textrender: Add support for ARGB and alignment properties
+ Fixes bug #581571.
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * ext/pango/gsttextrender.c:
+ textrender: Add ; after GST_BOILERPLATE to fix indention
-2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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.
+ vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
-2006-08-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
- * configure.ac:
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-gdp.xml:
- * gst/gdp/Makefile.am:
- * tests/check/Makefile.am:
- Move GDP plugin to -base from -bad. Closes #347783.
+ * gst/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.
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
* 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.
+ vorbistag: Store cover art in vorbiscomments
+ Fixes bug #513373.
- * tests/check/libs/tag.c: (GST_START_TEST):
- Some more tests for gst_tag_list_from_vorbiscomment_buffer().
+2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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).
+2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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
- * 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.
+2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2006-08-21 Wim Taymans <wim@fluendo.com>
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ Fix libdir for uninstalled gstreamer-app library
- * 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.
+2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- * gst-libs/gst/netbuffer/gstnetbuffer.h:
- Document GstNetBuffer.
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for APE tag caps
-2006-08-21 Stefan Kost <ensonic@users.sf.net>
+2009-05-12 01:35:27 +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
+ * configure.ac:
+ configure: bump core requirement to last release
+ as that's more likely to be true than that we need
+ only 0.21.1.
-2006-08-20 Stefan Kost <ensonic@users.sf.net>
+2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_unit_size), (set_structure_widths):
- Lower debug, use g_assert in _get_unit_size
+ * common:
+ * configure.ac:
+ configure: rename CVS -> git in a couple of places
- * 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
+2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-18 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ configure: bump GLib requirement to GLib >= 2.16
+ as per the New Regime (see wiki).
- * 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()
+2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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().
-
- * tests/check/libs/tag.c: (GST_START_TEST),
- (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
- More tests.
-
-2006-08-17 Wim Taymans <wim@fluendo.com>
+2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: support UYVY raw 4:2:2 in riff.
-2006-08-17 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * configure.ac:
+ Back to development -> 0.10.23.1
- * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
- (test_buffer_timestamps), (cddabasesrc_suite):
- Add unit test for the above.
+2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
- * tests/check/Makefile.am:
- Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
- to see what happens.
+ * ext/theora/theoradec.c:
+ theoradec: fix buffer overrun on 422 decode.
-2006-08-16 Wim Taymans <wim@fluendo.com>
+2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
- * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
- (gst_alsasink_open):
- * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
- (gst_alsasrc_open):
- Avoid setting and using a NULL device name.
- Print more info when we fail to open a device.
+ * ext/theora/theoradec.c:
+ theoradec: 444 support.
-2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
- * 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).
+ * ext/theora/theoradec.c:
+ theoradec: handle 422 images (as YUY2).
- * 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().
+2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
-2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theoradec: rearrange code in preparation for 422 and 444 support.
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
- Fix leak (#351502).
+=== release 0.10.23 ===
-2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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:
- * gst/playback/gstplaybin.c:
- Document playbin.
-
+ * 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-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-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 to CVS version.
-
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-08-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstplaybin.c:
- Update max volume to the same value that the volume element uses.
-
-2006-08-14 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Less uglyness..
-
-2006-08-14 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst/adder/gstadder.c:
- Don't clip float values. Fixes #350900.
-
-2006-08-11 Andy Wingo <wingo@pobox.com>
-
- * gst/tcp/gsttcp.c: Really fix the build?
-
- * gst/tcp/gsttcp.h: For now, always disable deprecation here --
- fixes the build.
-
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
- Float caps shouldn't have a "signed" field.
-
-2006-08-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-08-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-08-08 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-08-07 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-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):
- API: GstTheoraParse::synchronization-points
- Add a property 'synchronization-points' to fix badly synchronized oggs.
-
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Recognise ancient RealAudio files (see #349779).
-
-2006-08-03 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Jens Granseuer <jensgr at gmx net>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Add typefinder for Interplay's MVE format (#348973).
-
-2006-08-02 Wim Taymans <wim@fluendo.com>
-
- 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.
-
-2006-07-28 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-07-28 Wim Taymans <wim@fluendo.com>
-
- * gst/adder/gstadder.c: (gst_adder_setcaps),
- (gst_adder_query_duration):
- Fix leak in duration query.
- Reflow some docs and notes.
-
-2006-07-28 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-07-28 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-07-27 Wim Taymans <wim@fluendo.com>
-
- 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-07-24 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot 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.
-
-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.
-
- * tests/check/pipelines/vorbisenc.c:
- (test_discontinuity): New test, commented out until Mike lands
- some elite vorbisenc patches.
-
- * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
- Bufferstraw was actually factored out of these tests. Now we share
- code yay.
-
- * configure.ac (GST_MAJORMINOR): Rev core requirements to 0.10.9.1
- for bufferstraw addition to gstcheck.
-
-2006-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (clip_buffer):
- Better clipping.
-
-2006-07-21 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-07-20 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (cb_newpad), (main):
- Example of a small audio/video player using decodebin.
+ * gst-plugins-base.doap:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ Release 0.10.23
-2006-07-20 Stefan Kost <ensonic@users.sf.net>
+2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/riff/riff-ids.h:
- Add 'fact' chunk id
+ * 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
-2006-07-19 Wim Taymans <wim@fluendo.com>
+2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * 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
- * gst-libs/gst/rtp/gstbasertppayload.c:
- Add comment about better default MTU size.
+2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
- Small cleanups, start docs.
+ * 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.
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-07-18 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_fixate_channel_positions):
- Const-ify two arrays.
-
-2006-07-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-07-17 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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>
-
- * configure.ac:
- Bump nano back to CVS
-
-=== release 0.10.9 ===
-
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- releasing 0.10.9, "I walk the line"
-
-2006-07-14 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Fix missing g_unlock from the previous commit
-
-2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
- Reset the decoder in finalize so that all fields get cleared.
-
-2006-07-12 Wim Taymans <wim@fluendo.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
- basesink calculates silence sample correctly for us.
-
-2006-07-12 Wim Taymans <wim@fluendo.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-07-11 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
- (gst_ssa_parse_parse_line):
- Don't include the terminating NUL in the buffer size,
- it's only there for extra paranoia (would add random
- '*' characters at the end of each subtitle since the
- terminator itself is not valid UTF-8 technically).
- Also fix indenting after boilerplate macro.
-
-2006-07-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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 Wim Taymans <wim@fluendo.com>
-
- * 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 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).
-
-2006-07-06 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):
- Fix SMIL typefinding, make xml_check_first_element() more
- useful.
-
-2006-07-06 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-07-05 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/common/libgsttag.def:
- Export some new functions.
- * win32/vs6/libgstogg.dsp:
- Add a link to libgsttag-0.10.lib.
-
-2006-07-04 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
- Some const-ification.
-
-2006-07-04 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-07-03 Edward Hervey <edward@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
- Remove unused variable.
-
-2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am:
- include lcov.mak
- * configure.ac:
- add GCOV_LIBS to GST_LIBS
-
-2006-07-02 Tim-Philipp Müller <tim at centricular dot net>
-
- 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.
-
-2006-06-30 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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
-
-2006-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
- (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
- fix logging
-
-2006-06-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-06-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-06-22 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/tag.h:
- Oops, make that 'Since: 0.10.9'.
-
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/tag.h:
- * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
- (gst_tag_image_type_get_type):
- API: add GstTagImageType enum to describe images contained
- in image tags (#345641).
-
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-06-21 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-06-19 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
- (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
- (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
- (find_syncframe), (find_limits), (assign_value),
- (count_burst_unit), (gst_multi_fd_sink_new_client),
- (gst_multi_fd_sink_handle_client_write),
- (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
- (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
- (gst_multi_fd_sink_change_state):
- * gst/tcp/gstmultifdsink.h:
- Added shiny new burst-on-connect methods.
- Add properties to control the minimal amount of data queued.
- Small cleanups.
- API: bytes-min property
- API: time-min property
- API: buffers-min property
- API: burst-unit property
- API: burst-value property
- API: add-full signal
-
- * gst/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 Edward Hervey <edward@fluendo.com>
-
- * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
- Implement clipping for accurate seeking.
- Closes #345225
-
-2006-06-19 Wim Taymans <wim@fluendo.com>
-
- 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
-
-2006-06-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * configure.ac:
- Fix --disable-external (can't set conditionals conditionally,
- #343602).
-
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/elements/audioresample.c: (test_reuse),
- (audioresample_suite):
- Add test case for bug #342789 fixed below.
-
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
- * tests/check/elements/audioresample.c: (setup_audioresample),
- (cleanup_audioresample):
- Need to set element state here so that ::start and ::stop are
- called.
-
-2006-06-16 Wim Taymans <wim@fluendo.com>
-
- 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.
-
-2006-06-16 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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):
- Must dereference pointer to fourcc in the debug statement.
-
-2006-06-16 Stefan Kost <ensonic@users.sf.net>
-
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- add 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * autogen.sh:
- * configure.ac:
- * docs/Makefile.am:
- Use GST_PLUGIN_DOCS macro in configure.ac, add
- --enable-plugin-docs default to autogen.sh and use
- ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
-
-2006-06-15 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
- (gst_ogg_demux_loop):
- Combine GstFlowReturn from the source pads to give a
- meaningfull result to the upstream peer or to stop the
- processing task in case of errors.
-
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gststreaminfo.c: (cb_probe):
- Try GST_TAG_CODEC as fallback when extracting the
- codec name; more debug info.
-
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-06-14 Wim Taymans <wim@fluendo.com>
-
- Patch by: Alessandro Decina <alessandro at nnva dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
- (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
- (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
- (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
- Fix various leaks. Fixes #343699.
- Add x-smoke mime type.
-
-2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-ids.h:
- Add IDs for 'bext' chunks (see #343837).
-
-2006-06-12 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Young-Ho Cha <ganadist at chollian net>
-
- * gst/subparse/samiparse.c: (sami_context_pop_state),
- (handle_start_font), (end_sami_element):
- Honour font face tags in SAMI subtitles (#344503).
-
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
-
- * po/POTFILES.in:
- add missing files containing translatable strings
-
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
-
- * docs/libs/tmpl/.cvsignore:
- we don't want those *.sgml files in CVS either
-
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
-
- * docs/libs/.cvsignore:
- * tests/check/elements/.cvsignore:
- * tests/check/libs/.cvsignore:
- ignore more
-
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
-
- * docs/libs/Makefile.am:
- also commiting the changed Makefile.am (added more libs to the
- doc-build)
-
-2006-06-11 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- use GST_PKG_CHECK_MODULES, cleans up output
-
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-06-10 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- back to TRUNK
-
-=== release 0.10.8 ===
-
-2006-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.8, "Moar gij ziet mij nie"
-
-2006-06-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- * 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.
-
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-06-02 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-06-02 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-06-02 Michael Smith <msmith@fluendo.com>
-
- * gst/audioconvert/audioconvert.c: (float):
- More correct float->int conversion.
-
-2006-06-02 Michael Smith <msmith@fluendo.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.
-
-2006-06-02 Wim Taymans <wim@fluendo.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.
-
-2006-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-06-01 Stefan Kost <ensonic@users.sf.net>
-
- * 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 Stefan Kost <ensonic@users.sf.net>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * docs/libs/Makefile.am:
- set a magic variable to indicate we know the docs are incomplete
-
-2006-05-30 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/common/libgstvideo.def:
- export gst_video_calculate_display_ratio
- * win32/vs6/libgstvideoscale.dsp:
- add link to libgstvideo-0.10.lib
-
-2006-05-30 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-05-30 Wim Taymans <wim@fluendo.com>
-
- * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
- (gst_text_overlay_text_event):
- Added some FIXMEs.
-
-2006-05-30 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
- Fix build again.
-
-2006-05-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-05-30 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-05-30 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-05-29 Wim Taymans <wim@fluendo.com>
-
- * gst/adder/gstadder.c: (gst_adder_request_new_pad):
- Don't leak pad name.
-
-2006-05-29 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-05-29 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/elements/volume.c:
- make it compile again
-
-2006-05-28 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/elements/volume.c: (volume_suite):
- disable test until #343196 gets resolved
-
-2006-05-28 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
-
-2006-05-28 Edward Hervey <edward@fluendo.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * README:
- Replace current README (containing the release notes from
- some 0.9.x version) with a proper README taken from the core.
-
-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.
-
- * configure.ac:
- Bump requirements to core CVS (needed for vorbis
- typefinding to work).
-
-2006-05-24 Edward Hervey <edward@fluendo.com>
-
- * 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.
-
-2006-05-23 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-05-22 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-05-19 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
- * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
- 80 line columns
- Removed redundant floor()
-
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-05-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/elements/alsa.c: (test_device_property_probe):
- Fix test case: don't try to free NULL GValueArray when there
- are no devices.
-
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-05-18 Tim-Philipp Müller <tim at centricular dot 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/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>
-
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/subparse/samiparse.c: (handle_start_font):
- Don't ignore return value of strtol (++compiler_happiness).
-
-2006-05-17 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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.
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- Query supported sample rates. Fixes #341732.
-
-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 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
- (gst_alsasink_init):
- * ext/alsa/gstalsasink.h:
- Don't leak allocated snd_output_t structure if there's
- more than one alsasink instance at a time (#341873).
- Also fix GObject macros in header file.
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/subparse/gstsubparse.c:
- (gst_sub_parse_data_format_autodetect):
- Don't use libxml functions in the typefinding code.
-
-2006-05-15 Wim Taymans <wim@fluendo.com>
-
- * 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)
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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>
-
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
- (plugin_init):
- Add tentative support for libvisual-0.4 (#336881).
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Young-Ho Cha <ganadist at chollian net>
-
- * gst/subparse/samiparse.c: (handle_start_font):
- Need to map "silver" colour explicitly (#169936).
-
-2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-05-14 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-=== release 0.10.7 ===
-
-2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- releasing 0.10.7, "Leave the gun"
-
-2006-05-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
- Fix the build.
-
-2006-05-12 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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)
-
-2006-05-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/gstid3tag.c:
- Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
-
-2006-05-12 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/MANIFEST:
- update win32 files listing
-
-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
-
-2006-05-11 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-05-11 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-05-11 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-05-11 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * tests/check/elements/multifdsink.c: (wait_bytes_served),
- (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
- add two more tests, one doing streamheader
-
-2006-05-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-05-10 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * configure.ac:
- Require core CVS for the new newsegment stuff.
-
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Sjoerd Simons <sjoerd at luon net>
-
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
- Register nick for enum value (#341160).
-
-2006-05-09 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-05-09 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-05-09 Edward Hervey <edward@fluendo.com>
-
- * tests/icles/stress-xoverlay.c:
- Fix if core was built without parsing support.
-
-2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add SEDG (Samsung MPEG-4) fourcc.
-
-2006-05-09 Edward Hervey <edward@fluendo.com>
-
- * tests/icles/stress-xoverlay.c:
- * tests/examples/volume/volume.c:
- Fix if core was built without parsing support.
-
- * tests/examples/seek/seek.c:
- Disable the parse_launch example if core was built without parsing
- support.
-
-2006-05-08 Edward Hervey <edward@fluendo.com>
-
- * autogen.sh: (CONFIGURE_DEF_OPT):
- libtoolize on Darwin/MacOSX is called glibtoolize
-
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
-
- * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
- (adder_suite):
- Shuffle NULL state change around and raise timeout more
-
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
-
- * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
- (mp4_type_find), (plugin_init):
- Add typefind to distinguish between "audio/x-m4a" and new type
- "video/mp4". Fixes #340375
-
- * tests/check/elements/adder.c: (adder_suite):
- Raise timeout to make buildbot happy
-
-2006-05-07 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-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 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-05-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (cleanup_decodebin):
- Properly remove ghostpads. Fixes #340392
-
-2006-05-04 David Schleef <ds@schleef.org>
-
- * gst/typefind/gsttypefindfunctions.c:
-
-2006-05-03 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2006-05-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-04-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Protect against unexpected NULL strf_data buffer.
-
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- interpret the out[] buffer in the order the bytes are actually
- put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
- Other tests should use BYTE_ORDER since the array is filled in
- with actual values
-
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * tests/check/elements/audioconvert.c: (verify_convert),
- (GST_START_TEST):
- when a test fails, give an indication of which it is
-
-2006-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- compile fix; strtol() needs <stdlib.h>
-
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
-
- * 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/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
- (gst_adder_init):
- send events from src-pad to all sink-pads fixes #338657
-
-2006-04-28 Stefan Kost <ensonic@users.sf.net>
-
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
- (alsasink_parse_spec):
- query width capabilities from alsa, fixes #338919
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * 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 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
- Handle 0/1 framerate correctly Fixes #331901.
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_video_rate_event),
- (gst_video_rate_chain):
- Print more debugging info.
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
- (resample_set_state_from_caps):
- Add support for other formats audioresample can handle such as
- 32 bits in and float and 64 bits float. Fixes #301759
-
-2006-04-28 Wim Taymans <wim@fluendo.com>
-
- * gst/audioconvert/audioconvert.c: (float):
- correctly clip float samples > 1.0. Fixes #338718
-
-2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-04-28 Tim-Philipp Müller <tim at centricular dot 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.
-
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- back to HEAD
-
-=== release 0.10.6 ===
-
-2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.6, "Chao"
-
-2006-04-26 David Schleef <ds@schleef.org>
-
- * gst/videoscale/gstvideoscale.c: Add call to oil_init().
- Fixes #338897.
-
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * win32/common/config.h:
- new prerelease
-
-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.
-
-2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Edward Hervey <edward@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
- * tests/check/Makefile.am:
- * tests/check/elements/videorate.c: (assert_videorate_stats),
- (setup_videorate), (cleanup_videorate), (GST_START_TEST),
- (videorate_suite), (main):
- Fix an infinite loop if frames are passed in with wrongly ordered
- timestamps. Fixes #339013.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- fix typefinding on some ISO files. Fixes #339212.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- add another H264 fourcc. Fixes #339047.
-
-2006-04-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Patch by: Jan Schmidt
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_bufferalloc):
- Restore old StreamSelector behaviour.
- Fixes #338419.
-
-2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * win32/common/config.h:
- prerelease
-
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Antoine Tremblay <hexa00 at gmail dot com>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
- Fix some memory leaks: on finalize, free buffers left in the queue
- before destroying the queue; in _push(), unref rtp_buf even if
- the process vfunc returned a NULL buffer as output buffer (#337548);
- demote some recuring debug messages to LOG level.
-
-2006-04-11 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
- (gst_ogg_demux_loop):
- More cleanups.
- Respect segment stop when emiting EOS or SEGMENT_DONE.
- Fixes (#337945).
-
-2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_get_property):
- Don't leak pad name.
-
-2006-04-10 Michael Smith <msmith@fluendo.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst-plugins-base.supp:
- Suppress an old libtheora bug (fixed in more recent versions), so
- that FC4 buildslaves can pass.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
- (gst_ogg_demux_init), (gst_ogg_demux_finalize),
- (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
- (gst_ogg_demux_loop):
- Don't leak events.
- Remember what error we got when finding chains, if we
- were shutdown, that would not be an error.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event):
- Starting the ringbuffer when we did not acquire it can cause
- a deadlock, is pointless and causes nasty things for
- subclasses.
- Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
- (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_bisect_forward_serialno),
- (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
- Add some more debugging.
-
-2006-04-10 Wim Taymans <wim@fluendo.com>
-
- * ext/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.
-
-2006-04-09 Sebastien Moutte <sebastien@moutte.net>
-
- * gst/audioresample/debug.h:
- replace debug macros with variable number of parameters
- by a simple alias to gstreamer standard debug macros
- (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
- supported by MSVC 6.0 and 7.1)
- * gst/audioresample/resample.h:
- define M_PI and rint for WIN32
- * win32/common/libgstaudio.def:
- * win32/common/libgstriff.def:
- * win32/common/libgsttag.def:
- * win32/common/libgstvideo.def:
- add new exported functions
- * win32/vs6:
- update project files
-
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
-
- * ext/alsa/gstalsamixeroptions.c:
- (gst_alsa_mixer_options_class_init):
- * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_class_init):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init):
- * gst-libs/gst/audio/gstaudiosrc.c:
- (gst_audioringbuffer_class_init):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
- * gst-libs/gst/interfaces/colorbalancechannel.c:
- (gst_color_balance_channel_class_init):
- * gst-libs/gst/interfaces/mixeroptions.c:
- (gst_mixer_options_class_init):
- * gst-libs/gst/interfaces/mixertrack.c:
- (gst_mixer_track_class_init):
- * gst-libs/gst/interfaces/tunerchannel.c:
- (gst_tuner_channel_class_init):
- * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
- * gst-libs/gst/netbuffer/gstnetbuffer.c:
- (gst_netbuffer_class_init):
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init):
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_class_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
- * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
- * sys/v4l/gstv4lcolorbalance.c:
- (gst_v4l_color_balance_channel_class_init):
- * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
- (gst_v4l_tuner_norm_class_init):
- * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
- * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
- Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
-
-2006-04-08 Stefan Kost <ensonic@users.sf.net>
-
- * ext/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 Stefan Kost <ensonic@users.sf.net>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-04-07 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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. (Closes: #336617)
-
-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.
-
- * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
- buffers being pushed out. Fixes oggmux ! multifdsink.
-
-2006-04-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-04-04 Andy Wingo <wingo@pobox.com>
-
- * 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.
-
- * 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.
-
-2006-04-03 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/gsttheoradec.h:
- * ext/theora/theoradec.c: (theora_dec_src_event):
- Don't try to store a gdouble in a gboolean.
- Small cleanups.
-
-2006-04-03 Michael Smith <msmith@fluendo.com>
-
- * 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>
-
- * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
- a g_print.
-
- * 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.
-
- * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
-
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audiotestsrc/gstaudiotestsrc.c:
- Fixed the sample pipeline (see #323798)
-
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- * win32/common/config.h:
- * win32/common/config.h.in:
- use AS_VERSION and AS_NANO
- more cleanups
-
-2006-03-31 Andy Wingo <wingo@pobox.com>
-
- * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
- uninitialized variable return that would happen.
-
- * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
- uninitialized variable return that would never happen.
-
- * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
- (vorbis_parse_sink_event): Add an event function to flush our
- state on a seek, and to drain buffers on a premature EOS.
- (vorbis_parse_push_headers, vorbis_parse_clear_queue)
- (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
- (vorbis_parse_chain, vorbis_parse_queue_buffer)
- (vorbis_parse_drain_queue): Queue up buffers until we can set
- their timestamps and granulepos values.
-
- * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
- and keep track of data needed for deriving granulepos and
- timestamps for buffers.
-
-2006-03-29 Wim Taymans <wim@fluendo.com>
-
- 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
-
-2006-03-29 Wim Taymans <wim@fluendo.com>
-
- * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
- (stop_seek):
- Don't let double and triple clicks mess up our state.
-
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-03-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * tests/check/generic/clock-selection.c: (GST_START_TEST):
- set to NULL before unreffing, fixes a valgrind leak.
- Why was this not triggering the error that an object needs to
- be NULL before unreffing ?
- * win32/common/config.h:
- update
-
-2006-03-24 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-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
-
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
- * gst/adder/gstadder.c: (gst_adder_init):
- use DEBUG_FUNCPTR for collectpads
-
-2006-03-24 Stefan Kost <ensonic@users.sf.net>
-
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * ext/cdparanoia/gstcdparanoiasrc.c:
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
- (gst_gnome_vfs_sink_class_init):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
- * ext/ogg/gstoggmux.c:
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
- (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
- * ext/pango/gsttextoverlay.c:
- * ext/pango/gsttextrender.c:
- * ext/theora/theoradec.c:
- * ext/theora/theoraenc.c:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/audio/gstaudiofilter.c:
- (gst_audio_filter_base_init):
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- (gst_audio_filter_template_base_init):
- * gst/adder/gstadder.c: (gst_adder_get_type):
- * gst/adder/gstadder.h:
- * gst/audioconvert/gstaudioconvert.c:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
- (gst_audio_test_src_create):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/playback/gstdecodebin.c:
- * gst/playback/gstplaybin.c:
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_base_init):
- * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
- * gst/volume/gstvolume.c:
- * sys/v4l/gstv4lmjpegsink.c:
- * sys/v4l/gstv4lmjpegsrc.c:
- * tests/check/libs/cddabasesrc.c:
- * tests/old/examples/gob/gst-identity2.gob:
- Add docs for adder, use GST_ELEMENT_DETAILS macro,
- define GstElementDetails at the top
-
-2006-03-23 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/common/libgstinterfaces.def:
- Add a lot of export functions for gst-python
- * win32/vs6/libgstinterfaces.dsp:
- Add a missing include folder in the project configuration
-
-2006-03-23 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time), (gst_base_audio_src_create),
- (gst_base_audio_src_change_state):
- Fix audio sources, forgot to make the ringbuffer
- startable...
-
-2006-03-23 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time), (gst_base_audio_src_create),
- (gst_base_audio_src_change_state):
- unparent instead of unref the ringbuffer.
-
-2006-03-23 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- 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 Wim Taymans <wim@fluendo.com>
-
- 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 Edward Hervey <edward@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_dispose):
- Since we _parent the ringbuffer, we also need to
- _unparent instead of a plain _unref.
-
-2006-03-22 Wim Taymans <wim@fluendo.com>
-
- * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
- (stop_seek), (scrub_toggle_cb), (main):
- Add scrub checkbox.
-
-2006-03-21 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
- (gst_ogg_parse_chain):
- Fix very inefficient usage of linked lists (#335365).
-
-2006-03-21 Edward Hervey <edward@fluendo.com>
-
- * 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 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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>
-
- * 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.
-
-2006-03-17 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
- * win32/common/config.h:
- Revert accidental commits to these files.
-
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
-
- 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)
-
-2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/gstid3tag.c:
- Add TXXX frame identifiers for replaygain stuff as used
- by some taggers (see #323721).
-
-2006-03-16 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/tag/gstid3tag.c:
- Don't crash on unknown ID3v2 TXXX frames.
-
-2006-03-15 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
-
- 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.
-
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
-
- 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).
-
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for 24bpp DIB (#305279).
-
-2006-03-14 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * configure.ac:
- back to CVS.
-
-=== release 0.10.5 ===
-
-2006-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.5, "Net"
-
-2006-03-13 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/plugins/Makefile.am:
- Part of previous cdparanoiasrc docs fixes, forgot to commit.
-
-2006-03-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-03-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
- * gst-libs/gst/tag/tag.h:
- API: libgsttag: new method gst_tag_from_id3_user_tag().
-
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- NULL-terminate array of mpeg4 video file extensions.
- Fixes crash on PPC (#334226).
-
-2006-03-11 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnome_vfs_src_check_get_range):
- gnome_vfs_uri_is_local() alone is not a good indicator
- whether we can operate in pull-mode with a specific URI,
- as it returns FALSE for file:// URIs that point to an
- NFS-mounted path. Be more conservative here: whitelist
- local files, blacklist http URIs and use the old
- mechanism for anything else (fixes #334216).
-
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- back to trunk
-
-=== release 0.10.4 ===
-
-2006-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.4, "Power"
-
-2006-03-10 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-03-10 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Fabrizio Gennari
-
- * 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)
-
-2006-03-10 Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix for variable not initialized.
-
-2006-03-09 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (audiocast_register_listener), (gst_gnome_vfs_src_start):
- Some cleanups.
-
-2006-03-09 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
- Don't try to activate NULL chains.
-
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
- Fix invalid memory access to region before peek'd data (#332964).
-
-2006-03-09 Wim Taymans <wim@fluendo.com>
-
- Patch by: Christophe Fergeau
-
- * 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.
-
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
- Fix invalid memory access: make sure string passed to
- regexec() is NUL-termianted.
-
-2006-03-09 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-03-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-08 Michael Smith <msmith@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Make typefind function name for mpeg4 video unique.
-
-2006-03-08 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot 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.
-
-2006-03-08 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Fabrizio Gennari
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Allow palettes with less than 256 colours in AVI files
- (#333488)
-
-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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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().
-
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
- Don't ignore flow return from gst_pad_push().
-
-2006-03-07 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Fabrizio Gennari
-
- * 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)
-
-2006-03-06 Stefan Kost <ensonic@users.sf.net>
-
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- change location param details
- * gst/volume/gstvolume.c: (plugin_init):
- correct plugin description
-
-2006-03-06 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-03-06 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
- (gst_video_sink_class_init):
- Throw away frames that are later than 20 ms.
-
-2006-03-06 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Fabrizio Gennari
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Set depth on WMA caps (#333545)
-
-2006-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2006-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-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)
-
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Add support for '3IVD' fourcc (#333403).
-
-2006-03-04 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-03-04 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/vs6:
- add a project file for libgstvolume
- update the workspace
-
-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
-
-2006-03-02 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-03-02 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 Wim Taymans <wim@fluendo.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.
-
-2006-03-01 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/Makefile.am:
- Dist new header too,
-
-2006-03-01 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Sebastien Moutte <sebastien@moutte.net>
-
- * 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>
-
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
- Fix some docs.
-
-2006-02-28 Edward Hervey <edward@fluendo.com>
-
- * ext/alsa/gstalsasrc.c:
- Set proper class on the ElementDetails:
- Source/Audio instead of Src/Audio/
-
-2006-02-28 Edward Hervey <edward@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 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.
-
-2006-02-25 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
-
-2006-02-24 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2006-02-04 Michael Smith <msmith@fluendo.com>
-
- * 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).
-
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Fabrizio Gennari <fabrizio dot get at tiscali dot it>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
- Pick up palette for MS video v1 (#327028)
-
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-22 Wim Taymans <wim@fluendo.com>
-
- * ext/alsa/gstalsasrc.c:
- Fix description as pointed out by caugier.
-
-2006-02-22 Tommi Myöhänen <ext-tommi do myohanen at nokia dot com>
-
- Reviewed by : Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find):
- Better 3gp typefinding (#331526).
-
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-21 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/mixerutils.c:
- (element_factory_rank_compare_func):
- Make order in which elements are tried more determinable.
-
-2006-02-20 Julien MOUTTE <julien@moutte.net>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstdecodebin.c: (mimetype_is_raw):
- That should be text/x-pango-markup, not text/x-pango-layout.
-
-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.
-
-2006-02-17 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
- (gst_vorbisenc_convert_sink):
- Use some more scaling functions.
-
-2006-02-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-17 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-16 Tim-Philipp Müller <tim at centricular dot 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).
-
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-16 Mathieu Garcia <b0nk at free dot fr>
-
- 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.
-
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- Add gst-libs/gst/cdda to list of lib search paths.
-
-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.
-
-2006-02-15 Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- quicktime movie files can also contain 'uuid' atoms.
-
-2006-02-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-14 Wim Taymans <wim@fluendo.com>
-
- * 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
-
-2006-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-02-13 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-02-13 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-02-13 Tim-Philipp Müller <tim at centricular dot 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-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 Edward Hervey <edward@fluendo.com>
-
- * gst/playback/gstplaybasebin.c: (prepare_output):
- Remove stray semi-colon (fixes #330888).
-
-2006-02-12 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Added flv file typefind (video/x-flv).
-
-2006-02-10 Edward Hervey <edward@fluendo.com>
-
- * 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 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 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_setcaps), (gst_basertppayload_push):
- update seqnum before setting it on the packet; this makes sure
- that the timestamp and seqnum properties match after pushing
- a buffer
-
-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 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- Back to CVS
-
-=== release 0.10.3 ===
-
-2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- releasing 0.10.3, "Under Pressure"
-
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- Drat. Bump libtool version number for new API.
- Prelease 0.10.2.3 (of 0.10.3)
-
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
-
- * configure.ac:
- * win32/common/config.h:
- 0.10.2.2 prerelease (of 0.10.3).
-
-2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
- Revert Andy's newsegment change pending a more correct
- fix.
-
-2006-02-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
- (qt_type_find), (plugin_init):
- detect more files as 3gp
- group and reorder the iso file formats
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/vorbis/vorbis.c: (plugin_init):
- Register musicbrainz tags, so apps don't have to.
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-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.
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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).
-
-2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- Return FALSE from plugin_init() when GnomeVFS can't
- be initialised for some reason (#328423).
-
-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.
-
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-02-06 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
- * 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.
-
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst-libs/gst/tag/gstid3tag.c:
- Add mapping for ID3 International Standard Recording Code
- tag "TSRC"
-
-2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
- Don't leak tag names.
-
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/tag/gstid3tag.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- * gst-libs/gst/tag/tags.c:
- Split libgsttag docs into multiple sections.
-
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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.
-
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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.
-
-2006-02-05 Tim-Philipp Müller <tim at centricular dot 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?)
-
- * po/POTFILES.in:
- Add two translatable files.
-
-2006-02-05 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-05 Julien MOUTTE <julien@moutte.net>
-
- * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
- refactoring.
-
-2006-02-04 David Schleef <ds@schleef.org>
-
- * ext/ogg/gstoggmux.c:
- * gst/typefind/gsttypefindfunctions.c:
- Add Dirac typefinding and add dirac format to oggmux.
-
-2006-02-03 Tim-Philipp Müller <tim at centricular dot 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.
-
-2006-02-03 Edgard Lima <edgard.lima@indt.org.br>
-
- * ext/ogg/gstoggdemux.c:
- * ext/ogg/gstoggparse.c:
- * gst/tcp/gsttcpserversink.c:
- * sys/v4l/v4lsrc_calls.c:
- * sys/v4l/v4lsrc_calls.h:
- Just make it compile with --disable-gst-debug.
-
-2006-02-03 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_provide_clock):
- Ugh.. getting late I guess...
-
-2006-02-02 Wim Taymans <wim@fluendo.com>
-
- * 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstdecodebin.c: (try_to_link_1):
- Unlinking two source pads is ... hard.
-
-2006-02-02 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-02-01 Philippe Kalaf <burger at speedy dot org>
-
- 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.
-
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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.
-
-2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-
-2006-01-31 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (_theora_granule_time),
- (theora_dec_src_convert), (theora_dec_sink_convert):
- Call right _scale functions.
- Use parameter instead of some other random value.
-
-2006-01-31 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (_theora_granule_frame),
- (_theora_granule_time), (_inc_granulepos),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_handle_type_packet), (theora_handle_data_packet),
- (theora_dec_chain):
- Use higher precision timestamps calculation.
- Convert some other conversions to _scale.
-
-2006-01-31 Stefan Kost <ensonic@users.sf.net>
-
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audio_test_src_create_sine_table), (plugin_init):
- * gst/volume/gstvolume.c: (plugin_init):
- initialize gst_controller before using
-
-2006-01-31 Jan Schmidt <thaytan@mad.scientist.com>
-
- * tests/check/pipelines/theoraenc.c:
- * tests/check/pipelines/vorbisenc.c:
- Define constant using G_GINT64_CONSTANT to avoid errors when
- passing it around - otherwise it gets truncated to 32 bits.
-
- Fixes failing tests.
-
-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 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/vs7:
- add vs7 project files created by Sergey Scobich
-
-2006-01-30 Sebastien Moutte <sebastien@moutte.net>
-
- * win32/vs8:
- add vs8 project files created by Sergey Scobich
-
-2006-01-30 Andy Wingo <wingo@pobox.com>
-
- * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
- timestamp + duration, not just timestamp -- ogg pages should be
- ordered by stop time. Necessary fix given the change in vorbis
- timestamps.
-
- * ext/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 Thomas Vander Stichele <thomas at apestaart dot 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
-
-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.
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Andy Wingo <wingo@pobox.com>
-
- * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
-
- * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
- updated to timestamp from the first sample, not the last.
- (gst_vorbisenc_buffer_from_header_packet): New function, takes
- special care of granulepos and timestamp for header packets.
- (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
- when the first buffer has a nonzero timestamp.
-
- * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
- (GstVorbisEnc.subgranule_offset): New members. Take care of the
- case when the first audio buffer we get has a nonzero timestamp.
- (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
- properly timestamp vorbis buffers with the time of the first
- sample, not the last.
-
- * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
- vorbis_granule_time_copy -- now it takes the granule/subgranule
- offset into account.
-
- * tests/check/pipelines/vorbisenc.c: New test for correctness of
- timestamps, durations, and granulepos on buffers produced by
- vorbisenc.
-
-2006-01-30 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/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)
-
-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 Sebastien Moutte <sebastien@moutte.net>
-
- * 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>
-
- * 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
-
-2006-01-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * win32/MANIFEST:
- * win32/common/config.h:
- * win32/common/config.h.in:
- add config files just like in core
-
-2006-01-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
- Make gcc-4.1 happy (part of #327357).
-
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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.
-
-2006-01-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- check for and throw RESOURCE_BUSY
-
-2006-01-27 David Schleef <ds@schleef.org>
-
- * 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.
-
-2006-01-27 David Schleef <ds@schleef.org>
-
- * REQUIREMENTS:
- * configure.ac: update liboil requirement to 0.3.6
- * gst/videoscale/Makefile.am:
- * gst/videoscale/vs_scanline.c: liboilify
-
-2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-01-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2006-01-27 Julien MOUTTE <julien@moutte.net>
-
- * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
- (plugin_init): Make typefinding of subtitles work again.
-
-2006-01-26 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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 Edward Hervey <edward@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (pad_probe):
- Also consider the flush-start and tag events as unblockers
- for the pad probes.
-
-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 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-01-25 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2006-01-23 Edward Hervey <edward@fluendo.com>
-
- * 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 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 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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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>
-
- * 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.
-
-2006-01-19 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstplaybasebin.c: (setup_source):
- Free iterator when done.
-
-2006-01-17 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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
-
-=== release 0.10.2 ===
-
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.2, "Then the devil is six"
-
-2006-01-16 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_set_property):
- Comment out broken code that connects to the state-changed signal.
- At this point, changing current stream selection is broken, but
- stuff like gst-launch playbin current-audio=1 works and filters
- to the chosen stream.
-
-2006-01-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix #327216 (null dereference in vorbisdec)
-
-2006-01-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/theora/theoradec.c: (theora_handle_comment_packet):
- Post taglist actually on bus instead of just freeing it
- (fixes #327114 and totem bug #327080).
-
- * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
- Use gst_element_found_tags_for_pad(), so that the tags
- are sent downstream as an event as well.
-
-2006-01-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
- (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
- (gst_xvimagesink_buffer_alloc):
- move all regularly occurring messages to GST_LOG level
- add some more object logs
-
-2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
- fix a silly segfault
-
-2006-01-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/audio/mixerutils.c:
- * gst-libs/gst/audio/mixerutils.h:
- Add docs for mixerutils stuff.
-
-2006-01-13 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix playback for sources that emit raw audio or
- raw video streams (e.g.: cd audio sources) (#325984).
-
-2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/mixerutils.c:
- (gst_audio_mixer_filter_do_filter):
- actually save the element we create
-
-2006-01-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/cdda/gstcddabasesrc.c:
- (gst_cdda_base_src_handle_track_seek):
- No need to post a tag message on the bus when seeking
- within the same track, only post it when the current
- track changes.
-
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-01-11 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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.
-
-2006-01-11 Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
-
- 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).
-
-2006-01-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * tests/icles/Makefile.am:
- don't build the tests if we don't have the libs
-
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
- (gst_cd_paranoia_paranoia_callback):
- Don't try to free NULL pointers.
-
-2006-01-10 Edward Hervey <edward@fluendo.com>
-
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
- (gst_audio_rate_change_state), (plugin_init):
- Add debugging category.
- Fix type issues.
- Add case for incoming buffers without valid offset/offset_end.
-
-2006-01-10 Michael Smith <msmith@fluendo.com>
-
- * 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>
-
- * gst/playback/gstplaybin.c: (gen_audio_element):
- Don't leak an autoaudiosink/alsasink when we generate
- a new audio element. (old code, I guess)
-
-2006-01-10 Michael Smith <msmith@fluendo.com>
-
- * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
- Support float audio in audiorate.
- Use width rather than depth for selecting sample width.
-
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2006-01-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (wait_segment):
- * gst-libs/gst/audio/gstringbuffer.h:
- Name (private) union, makes Forte compiler happy (this time
- for real) (#324900).
-
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/Makefile.am:
- Link against libgstinterfaces, needed for mixer
- and property probe stuff.
-
-2006-01-09 Edward Hervey <edward@fluendo.com>
-
- * gst-libs/gst/Makefile.am:
-
-2006-01-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2006-01-03 Michael Smith <msmith@fluendo.com>
-
- * gst/audioresample/resample.h:
- As before, but for o_buf
-
-2006-01-03 Michael Smith <msmith@fluendo.com>
-
- * gst/audioresample/resample.h:
- Declare struct _ResampleState.buffer as unsigned char *, not void *,
- since we do arithmetic on it.
-
-2006-01-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
- (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
- (gst_ring_buffer_pause), (wait_segment):
- * gst-libs/gst/audio/gstringbuffer.h:
- Sun's Forte compiler doesn't seem to like anonymous structs,
- so use same setup as in GstBaseSrc (fixes #324900).
-
-2005-12-30 Stefan Kost <ensonic@users.sf.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
-
-2005-12-29 Stefan Kost <ensonic@users.sf.net>
-
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
- (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
- (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
- (gst_audio_test_src_create_square),
- (gst_audio_test_src_create_saw),
- (gst_audio_test_src_create_triangle),
- (gst_audio_test_src_create_silence),
- (gst_audio_test_src_create_white_noise),
- (gst_audio_test_src_create_pink_noise),
- (gst_audio_test_src_init_sine_table),
- (gst_audio_test_src_create_sine_table),
- (gst_audio_test_src_change_wave),
- (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
- (gst_audio_test_src_create), (gst_audio_test_src_set_property):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- update to basesrc changes, implement segmented seeking and eos
- handling, add a 'sine-tab' waveform for performance critical playback
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * po/POTFILES.in:
- ... and this time the other modified file that I missed last time.
-
-2005-12-29 Michael Smith <msmith@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (new_pad):
- Fix non-C89 variable declaration not at the start of a block. Should
- help some compilers.
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- And now fix 'make distcheck' (builddir != srcdir)
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/Makefile.am:
- Fix test case compilation.
-
-2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/gst-plugins-base-libs.types:
- Add docs for libgstcdda/GstCddaBaseSrc.
-
- * 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.
-
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2005-12-28 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/gnomevfs/gstgnomevfssink.h:
- Use GstBaseSinkClass as parent_class member for class struct, not
- GstBaseSink.
-
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Michael Benes
-
- * 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)
-
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2005-12-27 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-=== release 0.10.1 ===
-
-2005-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * configure.ac:
- releasing 0.10.1, "Dobro Dedek"
-
-2005-12-21 Edgard Lima <edgard.lima@indt.org.br>
-
- * gst/typefind/gsttypefindfunctions.c:
- iLBC30 and iLBC20 added to typefind.
-
-2005-12-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-2005-12-20 Edward Hervey <edward@fluendo.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.
-
-2005-12-19 Edward Hervey <edward@fluendo.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.
-
-2005-12-18 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.
-
- * 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
-
-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.
-
-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)
-
-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)
-
-2005-12-16 Stefan Kost <ensonic@users.sf.net>
-
- * 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
-
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
-
- * tests/check/generic/states.c: (GST_START_TEST):
- Blacklist cdparanoia element in state test.
-
-2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Benjamin Pineau
-
- * 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)
-
-2005-12-15 Michael Smith <msmith@fluendo.com>
-
- * 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.
-
-2005-12-15 Michael Smith <msmith@fluendo.com>
-
- * gst/audioresample/gstaudioresample.c:
- Don't leak all input buffers to audioresample.
-
-2005-12-15 Tim-Philipp Müller <tim at centricular dot 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.
-
-2005-12-15 Tim-Philipp Müller <tim at centricular dot 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/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 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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).
-
-2005-12-12 Christian Schaller <uraeus@gnome.org>
-
- * ext/Makefile.am: fix cdparanoia entry
- * gst-plugins-base.spec.in: add cdparanoia
-
-2005-12-12 Michael Smith <msmith@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Use the correct function to free list of typefind factories.
-
-2005-12-12 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 Wim Taymans <wim@fluendo.com>
-
- * 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 to disable flushing seeks.
- Disable scrubbing when doing non flushing seeks.
-
-2005-12-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-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.
-
-2005-12-09 Michael Smith <msmith@fluendo.com>
-
- * tools/Makefile.am:
- * tools/gst-launch-ext-m.m:
- Remove gst-launch-ext. It doesn't work, and is no longer
- particularly useful.
-
-2005-12-08 Luca Ognibene <luogni@tin.it>
-
- * 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.
-
-2005-12-07 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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>
-
- 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.
-
-2005-12-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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
-
-=== release 0.10.0 ===
-
-2005-12-05 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * configure.ac:
- releasing 0.10.0, "Mont-d'or"
-
-2005-12-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * tests/examples/seek/Makefile.am:
- Build fix for when gtk is not available.
-
-2005-12-05 Andy Wingo <wingo@pobox.com>
-
- * 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.
-
-2005-12-05 Andy Wingo <wingo@pobox.com>
-
- patch by: Kai Vehmanen <kv2004 eca cx>
-
- * gst-libs/gst/rtp/gstbasertpdepayload.c
- (gst_base_rtp_depayload_thread): Fix busy loop (#323017).
-
-2005-12-04 Andy Wingo <wingo@pobox.com>
-
- patch by: Sebastien Cote <sebas642 yahoo ca>
-
- * pkgconfig/gstreamer-plugins-base.pc.in (Libs): Add -L flag.
- Fixes #319172.
-
-2005-12-02 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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.
-
-2005-12-02 Wim Taymans <wim@fluendo.com>
-
- * 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.
-
-2005-12-02 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
- Small cleanups.
-
-2005-12-02 Wim Taymans <wim@fluendo.com>
-
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_transform):
- Post errors.
-
-=== release 0.9.7 ===
-
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
* configure.ac:
- releasing 0.9.7, "Mi Perro No Tiene Ninguna Nariz"
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am:
+ * 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:
- * win32/MANIFEST:
- * win32/gst.sln:
- add win32 MANIFEST file
- do something to the hungarian translation
-
-2005-12-01 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/Makefile.am:
- Add $(PANGO_DIR) to SUBDIRS
-
- * ext/pango/gstclockoverlay.c:
- * ext/pango/gsttimeoverlay.c:
- Fix and improve element descriptions.
-
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) 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
-
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * configure.ac:
- * ext/Makefile.am:
- moved pango to base
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot 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
-
-2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) 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
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * win32/:
- add Visual Studio 6 build files
-
-2005-12-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * 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
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ 0.10.22.5 pre-release
-2005-11-30 Wim Taymans <wim@fluendo.com>
+2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.h:
- * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
- (gst_basertppayload_event), (gst_basertppayload_push),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- closed #320644.
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcp-marshal.list:
+ multifdsink: fix signature of the add-full signal
+ The second parameter is a GstSyncMethod enum, not a boolean.
-2005-11-30 Julien MOUTTE <julien@moutte.net>
+2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * gst-libs/gst/video/gstvideofilter.c:
- * gst-libs/gst/video/gstvideosink.c:
- * gst-libs/gst/video/gstvideosink.h: Adding docs.
+ * gst/playback/gstplaysink.c:
+ playsink: initialize variable too
+
+2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
- * LICENSE:
- move
+ * configure.ac:
* po/af.po:
* po/az.po:
+ * po/bg.po:
+ * po/ca.po:
* po/cs.po:
+ * po/da.po:
+ * po/de.po:
* po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
* po/hu.po:
+ * po/id.po:
* po/it.po:
+ * po/ja.po:
+ * po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
- * Makefile.am:
- update
- * scripts/autoplugins.sh:
- remove
-
-2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * Makefile.am:
- * configure.ac:
- * examples/Makefile.am:
- * examples/capsfilter/Makefile.am:
- * examples/capsfilter/capsfilter1.c:
- * examples/gob/Makefile.am:
- * examples/gob/gst-identity2.gob:
- * examples/indexing/.cvsignore:
- * examples/indexing/Makefile.am:
- * examples/indexing/indexmpeg.c:
- * examples/seeking/.cvsignore:
- * examples/seeking/Makefile.am:
- * examples/seeking/cdparanoia.c:
- * examples/seeking/cdplayer.c:
- * examples/seeking/chained.c:
- * examples/seeking/scrubby.c:
- * examples/seeking/seek.c:
- * examples/stats/Makefile.am:
- * examples/stats/mp2ogg.c:
- * examples/switch/.cvsignore:
- * examples/switch/Makefile.am:
- * examples/switch/switcher.c:
- * tests/Makefile.am:
- * tests/check/generic/.cvsignore:
- * tests/check/pipelines/.cvsignore:
- * tests/examples/Makefile.am:
- * tests/examples/seek/Makefile.am:
- reorganize stuff under tests/
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
- Go away you stupid GstStaticPadTemplate memleak.
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/net/README:
- * gst-libs/gst/net/gstnetbuffer.c:
- * gst-libs/gst/net/gstnetbuffer.h:
- this was moved to "netbuffer"
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_get_type),
- (gst_video_filter_class_init), (gst_video_filter_init):
- * gst-libs/gst/video/gstvideofilter.h:
- borgify name to bring in line with other classes
-
-2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioscale/.cvsignore:
- * gst/audioscale/Makefile.am:
- * gst/audioscale/README:
- * gst/audioscale/audioscale.vcproj:
- * gst/audioscale/dtof.c:
- * gst/audioscale/dtos.c:
- * gst/audioscale/functable.c:
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/gstaudioscale.h:
- * gst/audioscale/private.h:
- * gst/audioscale/resample.c:
- * gst/audioscale/resample.h:
- * gst/audioscale/test.c:
- remove
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst-libs/gst/netbuffer/Makefile.am:
- really, really tired
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Update for new GstTypeFindFactory _register()
-
-2005-11-30 Edward Hervey <edward@fluendo.com>
-
- * gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
- Let's not override libgstnet from core for no reason...
- (libgstnetbuffer_@GST_MAJORMINOR@_la_SOURCES):
- Ok, maybe not so quick next time.
-
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ 0.10.22.4 pre-release
- * 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)
+2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
-2005-11-29 Andy Wingo <wingo@pobox.com>
+ * 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.
- * pkgconfig/gstreamer-plugins-base.pc.in:
- * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
- * gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
- (#322257).
+2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer size transformations
+ When calculating the input/output buffer sizes in the transform_size function,
+ take the number of channels into account, so we don't end up calculating
+ a buffer size that only contains a partial number of audio frames.
+ Also, when going from output size to input size, round down rather than
+ up, so as to calculate the minimum number of samples that *might* yield
+ a buffer of the intended destination size.
+ Fixes: #580470 and #580952
+
+2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/vorbis/gstvorbisenc.h:
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Ensure output buffers fall within the segment
+ Add the start position of the first segment to the running time
+ used to generate buffer timestamps in vorbisenc. This avoids generating
+ buffers which fall outside the initial segment. The element segment
+ handling requires more extensive fixing, but this at least prevents
+ regressions. Fixes: #580020
- * gst/playback/gststreamselector.c: (gst_stream_selector_chain):
- 3rd time's the charm. Correct ref-counting for discarded buffers.
+2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "add can-activate-pull property to baseaudiosink"
+ This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
- * 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
+2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
-2005-11-29 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ Revert "[baseaudiosink] add docs for can-activate-pull"
+ This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
- * gst/subparse/gstsubparse.c: (feed_textbuf):
- Don't access already unref'ed buffer.
+2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
-2005-11-29 Jan Schmidt <thaytan@mad.scientist.com>
+ [baseaudiosink] add docs for can-activate-pull
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
+ can-activate-pull.
- * 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
+2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
-2005-11-29 Julien MOUTTE <julien@moutte.net>
+ add can-activate-pull property to baseaudiosink
+ * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
+ to baseaudiosink.
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find): Free the list,
- patch from bug #322704 (Alessandro Decina).
+2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * gst-libs/gst/audio/Makefile.am:
- folded audiofilter into the audio library
+2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/gstvideoscale.c:
- remove unimplemented scale methods
+2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-11-28 Tim-Philipp Müller <tim at centricular dot net>
+ * 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).
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
- Don't leak caps.
+2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/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
- * 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 !
+2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
-2005-11-28 Edward Hervey <edward@fluendo.com>
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix unused variable in compilation with --disable-gst-debug
+ Fixes: #579668
- * 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.
+2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ Automatic update of common submodule
+ From b3941ea to 6ab11d1
- * 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
+2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-28 Wim Taymans <wim@fluendo.com>
+ * 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
- * gst/playback/gstplaybasebin.c: (group_commit), (probe_triggered):
- * gst/playback/gstplaybasebin.h:
- Prepare to handle errors betters.
+2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
+ Don't use REPLACE_ALL merge mode when that's not really what we want,
+ as now that REPLACE_ALL actually does what it's supposed to do in
+ core, we drop tags we wanted to keep, such as the various disc id
+ tags. Add unit test for this as well. Fixes #579463.
-2005-11-28 Wim Taymans <wim@fluendo.com>
+2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/audio/TODO:
- Updated TODO
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.16 API, we require only 2.14
+ Fixes #579267.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_open_device),
- (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
- (gst_audioringbuffer_release):
- Small cleanups.
+2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- Slave to the master clock when going to PLAYING and unslave when
- going to PAUSED.
-
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
- (gst_ring_buffer_acquire), (gst_ring_buffer_release),
- (gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
- (gst_ring_buffer_clear_all), (wait_segment),
- (gst_ring_buffer_commit), (gst_ring_buffer_read),
- (gst_ring_buffer_advance):
- * gst-libs/gst/audio/gstringbuffer.h:
- Add some docs and cleanups.
+ 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
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Fix navigation events
- coordinates translation with pixel aspect ratios.
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ RTCP: don't fail when retrieving invalid PT
+ We can't meaningfully assert on valid packet types so just return the type as it
+ is. Update the comments to reflect this.
+ Fixes #579192.
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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)
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.h:
+ * gst-libs/gst/app/gstappsrc.h:
+ app: add trivial cast macros
+ Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
+ and add the macros to the standard macros in the docs.
+ Fixes #579130
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ pkgconfig: add the app/ directory to Libs
+ Add the appsrc/appsink directory to the Libs in the uninstalled
+ pkgconfig file so that one can build against it.
+ Fixes #579129
-2005-11-28 Julien MOUTTE <julien@moutte.net>
+2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
- Refuse to create an XvImage if we can't find the format.
+ * configure.ac:
+ 0.10.22.2 pre-release
-2005-11-28 Edward Hervey <edward@fluendo.com>
+2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.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.
+ * ChangeLog:
+ ChangeLog: regenerate changelog with the gen-changelog script
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds):
- Don't try to create a zero-sized subbuffer.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/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
-2005-11-27 Julien MOUTTE <julien@moutte.net>
+2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.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:
+ * win32/common/_stdint.h:
+ * win32/common/config.h:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/video-enumtypes.c:
+ win32: Update win32 build files
-2005-11-27 Edward Hervey <edward@fluendo.com>
+2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (close_pad_link), (try_to_link_1):
- Remove unused properties, and add queues between demuxers and decoders
- so that a lot more files can preroll properly.
+ * tests/check/libs/video.c:
+ check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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
+ * 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
-2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * gst-libs/gst/video/video.h:
+ video: Fix typo in the docs
-2005-11-27 Julien MOUTTE <julien@moutte.net>
+2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Add support for YVYU YUV colorspace
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-plugins-base-libs-docs.sgml:
- * docs/libs/gst-plugins-base-libs-sections.txt:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstmixer.sgml:
- * docs/libs/tmpl/gstxoverlay.sgml: Add the XOverlay interface
- to documentation.
+ * gst-libs/gst/fft/gstfft.c:
+ docs: fix hyperlink and move fft attribution to the right place
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * sys/xvimage/xvimagesink.c: Added doc for xvimagesink.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ log: use G_GUINT64_FORMAT instead of llu
-2005-11-26 Julien MOUTTE <julien@moutte.net>
+2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-adder.xml:
- * docs/plugins/inspect/plugin-alsa.xml:
- * docs/plugins/inspect/plugin-audioconvert.xml:
- * docs/plugins/inspect/plugin-audiorate.xml:
- * docs/plugins/inspect/plugin-audioresample.xml:
- * docs/plugins/inspect/plugin-audiotestsrc.xml:
- * docs/plugins/inspect/plugin-decodebin.xml:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/inspect/plugin-gnomevfs.xml:
- * docs/plugins/inspect/plugin-ogg.xml:
- * docs/plugins/inspect/plugin-playbin.xml:
- * docs/plugins/inspect/plugin-subparse.xml:
- * docs/plugins/inspect/plugin-tcp.xml:
- * docs/plugins/inspect/plugin-theora.xml:
- * docs/plugins/inspect/plugin-typefindfunctions.xml:
- * docs/plugins/inspect/plugin-video4linux.xml:
- * docs/plugins/inspect/plugin-videorate.xml:
- * docs/plugins/inspect/plugin-videoscale.xml:
- * docs/plugins/inspect/plugin-videotestsrc.xml:
- * docs/plugins/inspect/plugin-volume.xml:
- * docs/plugins/inspect/plugin-vorbis.xml:
- * docs/plugins/inspect/plugin-ximagesink.xml:
- * docs/plugins/inspect/plugin-xvimagesink.xml: Updated version.
- * sys/ximage/ximagesink.c: Adding documentation for ximagesink.
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ RTSP: add missing headers for WMS RTSP
+ Add missing headers related to Windows Media RTSP extension.
+ Fixes #578942
-2005-11-26 Edward Hervey <edward@fluendo.com>
+2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
+
+ * 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/audiorate/gstaudiorate.c: (gst_audiorate_chain):
- Properly return GstFlowReturn from gst_pad_push in chain functions.
+2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
-2005-11-25 Michael Smith <msmith@fluendo.com>
+ * ext/theora/theoraenc.c:
+ theoraenc: factor out keyframe forcing
+ See #578656
- * 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.
+2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-24 Andy Wingo <wingo@pobox.com>
+ * AUTHORS:
+ * gst-libs/gst/fft/gstfft.c:
+ Give credit to Mark Borgerding (kissfft author)
+ and add myself to AUTHORS as well. Fixes #575638.
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_handle_client_write):
- Be threadsafe.
+2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * configure.ac: back to HEAD
+2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.9.6 ===
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ multifdsink: add property to handle client write
+ Add a property to disable listening to client writes. This property is usefull
+ when other code will deal with reading from the client socket.
+ API: GstMultiFdSink::handle-read property
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
- * configure.ac:
- releasing 0.9.6, "White Eight"
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ * win32/common/libgstrtp.def:
+ RTCP: add beginnings of Feedback messages
+ Add the beginnings of parsing and constructing Feedback messages.
+ Fixes #577610.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- * docs/plugins/inspect/plugin-sine.xml:
- remove sinesrc some more
+ * gst/playback/gstplaysink.c:
+ playbin2: clear the target
+ Clear the target of our ghostpads before we remove the pad from the element.
+ This to make sure that the internal pad is not left linked to whatever pad we
+ were ghosted to. This should only be a problem when we leak the ghostpads.
+ Also release our subpicture pads.
+ Fixes #577288.
-2005-11-23 Wim Taymans <wim@fluendo.com>
+2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
- If we are reading too slowly, jump forward in the ringbuffer
- instead of blocking.
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
+ Fixes #570768.
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/libvisual/visual.c: (gst_visual_src_setcaps), (get_buffer),
- (gst_visual_chain):
- * ext/theora/theoraenc.c: (theora_enc_sink_setcaps):
- * gst/videorate/gstvideorate.c: (gst_videorate_setcaps),
- (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate), (gst_videotestsrc_create):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get_fps),
- (gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query),
- (gst_v4lmjpegsrc_get), (gst_v4lmjpegsrc_getcaps):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps),
- (gst_v4lsrc_get_fps_list), (gst_v4lsrc_buffer_new):
- Updates for API changes
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: adjust the internal timestamp
+ Adjust the internal timestamp before comparing it against the adjusted clock
+ time.
+ Fixes #578506
-2005-11-23 Wim Taymans <wim@fluendo.com>
+2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
- (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
- Fix for calibration API change.
+ 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
-2005-11-23 Michael Smith <msmith@fluendo.com>
+2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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_*().
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * win32/common/libgstaudio.def:
+ audioclock: add methods for the internal offset
+ Add two methods for getting the unadjusted time of the clock and one for
+ adjusting an internal time. We will need these methods for correctly handling
+ the time after a gst_audio_clock_reset().
+ Add a debug category and some debug lines to the audio clock.
+ API: gst_audio_clock_get_time()
+ API: gst_audio_clock_adjust()
+ API: GST_AUDIO_CLOCK_CAST()
-2005-11-23 Edward Hervey <edward@fluendo.com>
+2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * autogen.sh:
- eheh, sinesrc is gone. Replacing sample srcfile by existing one.
- Fixes autogen
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: fix up the debugs and warnings
+ Use _OBJECT variants because we can. Go over some log statements and put them in
+ the right category.
+ Fixes #567740.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
- * check/Makefile.am:
- * check/elements/videotestsrc.c: (setup_videotestsrc),
- (cleanup_videotestsrc), (GST_START_TEST), (videotestsrc_suite),
- (main):
- add a test for videotestsrc
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: fix error in sync-method
+ Multifdsink did not handle sync-method=latest-keyframe correctly when the
+ soft-limit is set to -1 (unlimited).
+ Fixes #578583.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/.cvsignore:
- * gst/sine/Makefile.am:
- * gst/sine/gstsinesrc.c:
- * gst/sine/gstsinesrc.h:
- * gst/sine/sinesrc.vcproj:
- and remove sinesrc from the repository. Closes #321446
+ * 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.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- * gst-plugins-base.spec.in:
- remove sinesrc from the build
+ * tests/examples/seek/seek.c:
+ seek: handle clock-lost messages
+ When we receive a clock-lost message we need to pause and play to select a new
+ clock.
-2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * tests/check/Makefile.am:
+ * tests/check/elements/playbin2.c:
+ check: add a unit test for playbin2
+ Add unit test for playbin2 and include the refcount test in #577794.
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
+2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * gst/playback/gstplaysink.c:
+ playbin2: fix refcounting of visualisations
+ See #577794.
- * 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.
+2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-23 Jan Schmidt <thaytan@mad.scientist.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.
- * ext/libvisual/visual.c: (gst_visual_chain):
- Fix the fps calculations.
+2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/avcodec.h:
- Move structure element for clarity
+ * tests/check/elements/appsink.c:
+ check: fix appsink test
+ Fix the appsink test now that the method signature changed.
- * 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...
+2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-22 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: handle missing input-selector
+ Gracefully degrade and disable stream selection when input-selector is
+ missing.
- * 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...
+2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
-2005-11-22 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ appsink: make callbacks return GstFlowReturn
+ Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
+ errors can be reported properly.
+ Fixes #577827.
- * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
- This clock can be slaved to a master clock now.
+2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
- (gst_base_audio_sink_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/audio/gstringbuffer.c:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ ringbuffer: allow for custom commit functions
+ Allow subclasses to override the commit method.
- * 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.
+2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * 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
+2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
-2005-11-22 Michael Smith <msmith@fluendo.com>
+ * 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.
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- Add underscore.
+2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
-2005-11-22 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ audiofilter: don't leak pad-template
+ gst_element_class_add_pad_template() does not take ownership.
- * gst/videorate/gstvideorate.c: (gst_videorate_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- Use utility method for scaling clocktime for fractional framerates.
+2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
-2005-11-22 Jan Schmidt <thaytan@mad.scientist.com>
+ * common:
+ Automatic update of common submodule
+ From d0ea89e to b3941ea
- * 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.
+2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/audio/audio.h:
- remove some deprecated functions
+2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-11-22 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoradec.c:
+ theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
+ This fixes most seeking issues when used with gnonlin.
+ Fixes #543591
- * Update for gst_tag_setter API changes.
+2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
-2005-11-22 Andy Wingo <wingo@pobox.com>
+ * common:
+ Automatic update of common submodule
+ From f8b3d91 to d0ea89e
- * 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.
+2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
-2005-11-22 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't leak selector when getting current stream numbers.
- * examples/seeking/seek.c: (main):
- Give higher priority to bus signals than the gtk events
- to fix a race condition in the segment looping.
+2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-plugins-base.spec.in:
- Rename libgsttagedit to libgsttag (#322117).
+2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-22 Tim-Philipp Müller <tim at centricular 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.
- * 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.
+2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-11-22 Julien MOUTTE <julien@moutte.net>
+ * 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.
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event): Handle navigation
- events correcly with borders if applicable.
+2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
-2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
+ * tests/check/elements/.gitignore:
+ ignores: Ignore the videoscale check binary
- Patch by: Luca Ognibene
+2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
- * 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
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_set_proxy to the win32 exports
-2005-11-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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
+ * ext/alsa/gstalsamixer.c:
+ alsamixer: don't forget to release locks in a few places
+ Might fix #576585.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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).
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't read over line ends when taking the last Cr or Cb
- * sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
- (gst_xvimagesink_xoverlay_init):
- Remove desired size stuff (#321816).
+2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Don't write to few pixels and don't mix Cr and Cb
+ Fixes bug #577054.
- * gst/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.
+2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-21 Andy Wingo <wingo@pobox.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.
- 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.
+2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-21 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstplaybin2.c:
+ docs: add a blurb about redirect messages to playbin2 docs
- * gst/playback/gstplaybin.c (gen_audio_element)
- (gen_video_element): Use the new MISSING_PLUGIN core error
- category. Closes #320060.
+2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix little typo in the comments
- * gst/subparse/gstsubparse.c (gst_subparse_src_event):
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek): Update for
- stream lock changes.
+2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-21 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-21 Andy Wingo <wingo@pobox.com>
+ * 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.
- * *.h:
- * *.c: Ran scripts/update-macros. Oh yes.
+2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ docs: fix typo in gst_rtsp_message_unset() API docs
- * 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).
+2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-21 Michael Smith <msmith@fluendo.com>
+ * 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.
- * 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.
+2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
+ This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
- * 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)
+2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
+ According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
+ format the colorkey depending on xcontext->depth. This is what they will use to
+ interprete the value. The max_value in turn is usualy a constant regardless of
+ the depth.
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: reset whole message (was sizeof pointer instead of sizeof type)
-2005-11-21 Andy Wingo <wingo@pobox.com>
+2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
- patch by: Alessandro Dessina <alessandro nnva org>
+ * gst-libs/gst/interfaces/mixer.c:
+ doc: Fix a typo in the GstMixer docs
- * 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.
+2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Fix linear scaling for one byte components
+ Fixes bug #577054.
- * 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.
+2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-21 Michael Smith <msmith@fluendo.com>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Fix 4tap scaling of YUYV and friends
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
- (gst_vorbisenc_chain), (gst_vorbisenc_output_buffers):
- Properly handle pad_push return values.
+2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-20 Jan Schmidt <thaytan@mad.scientist.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.
- * 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>
+2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Fix the audiosrc base class again, we did not unflush.
+ * tests/check/elements/videoscale.c:
+ videoscale: Add some more unit tests
-2005-11-17 Julien MOUTTE <julien@moutte.net>
+2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Use bilinear instead of 4tap scaling for heights < 4
+ Partially fixes bug #577054.
-2005-11-17 Wim Taymans <wim@fluendo.com>
+2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * 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.
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_configure_segment),
- (gst_ogg_demux_perform_seek):
- Segment done must include stream time.
+ * gst/videoscale/vs_scanline.c:
+ videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
+ Partially fixes bug #577054.
- * ext/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.
+2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ * 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).
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init):
- Fix compile again.
+2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * tests/check/Makefile.am:
+ * tests/check/elements/videoscale.c:
+ videoscale: Add a lot of unit tests
- * 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
+2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videoscale/gstvideoscale.c:
+ videocale: Add support for video/x-raw-gray with bpp=depth=8
- * check/generic/states.c: (GST_START_TEST):
- fix the test so that it only checks for elements that are part of
- this source module
+2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Michael Smith <msmith@fluendo.com>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
- * 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.
+2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Edward Hervey <edward@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_event), (gst_videorate_chain):
- Handle segment seeks
+2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-16 Wim Taymans <wim@fluendo.com>
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
- * 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.
+2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Add support for v308 YUV colorspace
-2005-11-16 Julien MOUTTE <julien@moutte.net>
+2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_center_rect):
- * gst-libs/gst/video/videosink.h: Add helper function needed
- for video sinks.
+ * gst/videoscale/vs_4tap.c:
+ videoscale: Add my copyright to the 4tap scalers
-2005-11-16 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- Don't leak reference to pad parent.
+ videoscale: Enable 4-tap scaling for all supported formats
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB565 and RGB555
-2005-11-16 Wim Taymans <wim@fluendo.com>
+2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * examples/seeking/seek.c: (send_event), (do_seek),
- (loop_toggle_cb), (segment_done), (main):
- Added looping checkbox.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for UYVY
-2005-11-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for YUY2 and YVYU
-2005-11-15 Johan Dahlin <johan@gnome.org>
+2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGB and BGR
-2005-11-15 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
- Make state change function thread safe.
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
-2005-11-15 Edward Hervey <edward@fluendo.com>
+2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: Fix drawing of UYVY text borders
-2005-11-14 Julien MOUTTE <julien@moutte.net>
+2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
- * 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.
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ textoverlay: Add support for UYVY colorspace
+ Fixes bug #378094.
-2005-11-14 Michael Smith <msmith@fluendo.com>
+2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_client_queue_data),
- (gst_multifdsink_render):
- Don't leak GDP headers when using GDP mode (i.e. tcpserversink).
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: do some more cleanup
+ Free the groups when we go to READY.
+ Allow for NO_PREROLL elements.
-2005-11-14 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: start CSeq counting from 1 instead of 0
+ Start counting from 1 instead of 0 as this is what most other clients
+ seem to do.
- * gst/playback/gstplaybin.c (gen_audio_element): Use autoaudiosink
- as well if it is available. Fixes #316442.
+2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-14 Michael Smith <msmith@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add ETag and If-Match headers
+ Add new headers, we need them for RealMedia support.
- * 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.
+2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-11-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- * 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
+2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-11-13 Edward Hervey <edward@fluendo.com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ audiosrc: improve 'Dropped n samples' warning message
- * 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.
+2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * configure.ac: back to HEAD
+2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-=== release 0.9.5 ===
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: add more support for subpictures
-2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- releasing 0.9.5, "No No Kia"
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playbin2: first support for subpictures
+ Add beginnings of subpicture support.
-2005-11-11 Edward Hervey <edward@fluendo.com>
+2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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"
+ * tests/examples/seek/seek.c:
+ seek: print tags from the different tracks
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Check whether peer pad exists before sending navigation events
- to it.
+ * gst/playback/gstplaybin2.c:
+ playbin2: blacklist subpictures for now
+ Blacklist the subpictures until we add support for them.
+ Add some small debug info.
+ See #576408.
-2005-11-11 Michael Smith <msmith@fluendo.com>
+2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-2005-11-11 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/playback/gstplaysink.c:
+ playbin2: rescan audio sinks for volume/mute
+ Rescan the audio sinks for the mute and volume properties.
+ fixes #576180.
-2005-11-10 Andy Wingo <wingo@pobox.com>
+2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c (prepare_output): Fix format
- string doober.
+ * 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
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps):
- Prefer passthrough in transform_caps
+ * 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
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2005-11-10 Wim Taymans <wim@fluendo.com>
+2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-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.
+2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-11-10 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/README:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ docs: add examples for tcp elements, also use correct section name. Fixes #564139
+ Updated the examples in the README to actually work. Add them to api docs. Tests
+ the api-docs and fix the section names to make the docs actualy show up.
+ The example for "tcpserversrc" needs review (might be an element bug).
- * 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.
+2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-11-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videoscale/gstvideoscale.c:
+ indent: fix damange that gst-indent did some time ago
- * 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
+2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-10 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.
- * 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.
+2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-10 Michael Smith <msmith@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
- * 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.
+2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-11-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose):
- Set elements to NULL state before disposing of them.
+2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
-2005-11-08 Sebastien Cote <sebas642 at yahoo dot ca>
+ * ext/vorbis/vorbisenc.c:
+ vorbisenc: Set caps on non-header ouput buffers.
+ Fixes #576142.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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>
+ * tests/examples/seek/seek.c:
+ seek: Add some more debug
+ Add some more info about the selected streams.
- * gst/audioconvert/gstaudioconvert.c:
- Fix typo in docs.
+2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: a pad starts out being not drained.
+ Mark a new pad as not drained until we get EOS on it.
- * 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.
+2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
-2005-11-08 Brian Cameron <brian dot cameron at sun dot com>
+ * gst/playback/gstqueue2.c:
+ win32: fix seeking in large files
+ Fix Seeking in large files by using the 64-bit seek functions.
+ Fixes #576019
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/volume/gstvolume.c: (volume_set_caps):
- Fix compilation on Solaris with Forte. (#320923)
+ * 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.
-2005-11-08 Wim Taymans <wim@fluendo.com>
+2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- No need to do a typecheck.
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: more cleanups and docs.
+ Add some more comments and use g_list_prepend().
-2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/alsa/gstalsa.h:
- We register a debug category, so let's use it.
+ * 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.
-2005-11-04 Zeeshan Ali <zeenix@gmail.com>
+2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_wait):
- Fixed a small problem.
+ * 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.
-2005-11-04 Wim Taymans <wim@fluendo.com>
+2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * tests/icles/playbin-text.c:
+ icles: print better error and warning messages
+ --
- * 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.
+2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-11-04 Zeeshan Ali <zeenix@gmail.com>
+ * 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/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.
+2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-03 Wim Taymans <wim@fluendo.com>
+ * 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.
- * 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.
+2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-11-02 Sebastien Cote <sebas642@yahoo.ca>
+ * 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.
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push),
- (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>
+ * 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.
- * 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.
+2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+ * 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.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_add_to_queue),
- (gst_base_rtp_depayload_push):
- Fixed the queueing algorithm.
+2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-31 Zeeshan Ali <zeenix@gmail.com>
+ * gst/playback/gstplaysink.c:
+ playbin: set custom text sink properties
+ Set the custom sink async=FALSE to not make it participate in preroll because we
+ are dealing with sparse streams.
+ Try to set sync=TRUE on the custom text sink.
- * gst-libs/gst/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_push):
- A small fix
+2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-31 Wim Taymans <wim@fluendo.com>
+ * tests/icles/playbin-text.c:
+ example: use appsink instead of fakesink
+ Use appsink instead of fakesink to get the subtitles.
+ Make things more pretty.
- * gst-libs/gst/audio/gstringbuffer.h:
- Don't break ABI.
+2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2005-10-31 Wim Taymans <wim@fluendo.com>
+2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * tests/check/elements/appsink.c:
+ tests: fix include in the appsink test
+ Fix dist by doing the right include.
-2005-10-29 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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)
+ * gst/playback/gstplaybin2.c:
+ playbin2: don't try to set invalid stream numbers
+ Fix a problem with setting the stream numbers because we check for the wrong
+ range.
+ See #575239.
-2005-10-28 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/gstplaybin2.c:
+ playbin2: release the shutdown lock
+ Release the shutdown lock when we wait for other groups to complete or else we
+ have a deadlock when the other group completes and tries to grab the shutdown
+ lock.
+ Fixes #575550.
-2005-10-27 Philippe Khalaf <burger at speedy dot org>
+2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/rtp/gstbasedepayload.c:
- * gst-libs/gst/rtp/gstbasedepayload.h:
- Minor cleanups
+ * tests/examples/app/appsrc-ra.c:
+ * tests/examples/app/appsrc-seekable.c:
+ * tests/examples/app/appsrc-stream.c:
+ * tests/examples/app/appsrc-stream2.c:
+ examples: fix g_object_set() value type.
+ Make sure we cast the length value as a gint64 to the vararg g_object_set() just
+ incase sizeof(gsize) != sizeof(gint64).
-2005-10-27 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/.cvsignore:
- * gst/playback/decodetest.c:
- * gst/playback/test3.c:
- Port these two tests as well.
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: make flac typefinder return lower probability for frame headers
+ The flac frame header typefinder overstates the likelihood of a match, leading
+ to false positives with e.g. aac streams and PDF files. Reduce probabilty
+ returned from LIKELY to POSSIBLE for the frame header matchin code.
+ Fixes #574939.
-2005-10-27 Wim Taymans <wim@fluendo.com>
+2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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/typefind/gsttypefindfunctions.c:
+ typefinding: improve image/bmp typefinder
+ Detect more variations and also bail out in more cases where the values
+ don't make sense. Furthermore, add width/height and bpp to the caps,
+ because we can.
-2005-10-27 Wim Taymans <wim@fluendo.com>
+2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
- (gst_adder_change_state):
- Fix timestamps and fix deadlock when stopping the collectpads.
+ * tests/check/Makefile.am:
+ check: Ignore alsamixer in the states test too
-2005-10-27 Zeeshan Ali <zeenix@gmail.com>
+2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/rtp/gstrtpbuffer.h:
- Declaring the payload types as strings too so that they can be used
- in the padtemplate inialization.
+ * sys/v4l/v4l_calls.c:
+ v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
-2005-10-26 Andre Magalhaes <andrunko@gmail.com>
+2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- reviewed by: Zeeshan Ali <zeenix@gmail.com>
+ * 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/rtp/gstbasertpdepayload.c:
- (gst_base_rtp_depayload_class_init):
- Fixes a small but nasty bug. The derived elements no longer segfaults
- on finalization.
+2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-26 Michael Smith <msmith@fluendo.com>
+ * ext/vorbis/vorbisparse.c:
+ vorbisparse: be smarter when queueing headers
+ Look at the first buffer byte to see if a buffer is a header instead of counting
+ packets.
- * gst/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.
+2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-26 Zeeshan Ali <zeenix@gmail.com>
+ * 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.
- * 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.
+2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-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
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: add some debug info
+ Add some debug info to log when the seek worked.
-2005-10-26 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2005-10-25 Zeeshan Ali <zeenix@gmail.com>
+2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * ext/vorbis/vorbisdec.c:
+ vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
-2005-10-24 Wim Taymans <wim@fluendo.com>
+2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * ext/theora/theoradec.c:
+ theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
-2005-10-24 Julien MOUTTE <julien@moutte.net>
+2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Check
- the caps against our xcontext caps.
+ * gst/playback/gsturidecodebin.c:
+ playbin2: fix raw elements like cdda://
+ Fix a fixme with a one liner and make cd playback work again.
-2005-10-24 Wim Taymans <wim@fluendo.com>
+2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2005-10-24 Wim Taymans <wim@fluendo.com>
+2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
- * 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.
+ * gst-libs/gst/tag/gstvorbistag.c:
+ vorbistag: Protect memory allocation calculation from overflow.
+ Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
-2005-10-24 Julien MOUTTE <julien@moutte.net>
+2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
- * 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.
+ * gst-plugins-base.spec.in:
+ Spec: fix up deps
-2005-10-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- back to HEAD
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix parsing of the timeout parameter
+ --
-=== release 0.9.4 ===
+2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: fix g_return condition
+ when parsing a data message, we require a data message.
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.9.4, "Velociraptor"
+2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- * po/POTFILES.in:
- STOPPED -> FAILED
+2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-21 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: free the right string.
+ Free the key value before we remove the header item from the array. The item we
+ retrieved from the array is only valid until we remove it from the array.
- * 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.
+2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-21 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: keep track of amount of decoded bytes
+ Keep track of the actual amount of decoded bytes, which can be less than 3 when
+ we decode the last bits of a base64 message.
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisdec.h:
- Fix old naming.
+2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Don't try to sync on buffers without a timestamp.
+ * gst/adder/gstadder.c:
+ adder: log details in getcaps like in setcaps
-2005-10-21 Wim Taymans <wim@fluendo.com>
+2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_sink_event):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisdec.h:
- Fix old naming.
+ * win32/MANIFEST:
+ win32: update MANIFEST, fixing 'make dist'
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_query_types),
- (gst_vorbisenc_src_query):
- Implement position and duration queries.
+ * common:
+ Automatic update of common submodule
+ From 7032163 to f8b3d91
- * gst/playback/test3.c: (update_scale), (main):
- Fix for async state changes and print nicer output.
+2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
-2005-10-20 Wim Taymans <wim@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: add photoshop typefind functions
+ Add photoshop typefind functions.
+ Fixes #574516.
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- * gst/playback/test6.c: (main):
- Fix tests again
+2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
+ * 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.
- * 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.
+2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-20 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ rtsp: use gstreamer dump mem
+ --
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gst_play_base_bin_change_state):
- Fix leak.
- Handle case where playbasebin is now ASYNC because
- decodebin is.
+2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-19 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: use glib base64 encoder
+ --
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/bufferframesconvert.c:
- * gst/audioconvert/plugin.c: (plugin_init):
- * gst/audioconvert/plugin.h:
- And bye bye buffer-frames-convert
+2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-10-19 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstdecodebin2.c:
+ Unblock blocked ghostpads when shutting down. Fixes #574293.
- * 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>
+2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add mapping for Fraps video codec.
+ Found through insanity testrun. Confirmed mapping in libavformat.
- * 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.
+2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-17 Stefan Kost <ensonic@users.sf.net>
+ * 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.
- * gst/audiotestsrc/Makefile.am:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- fix broken build of controllerized plugins
+2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-17 Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: Use the proper data pointer instead of poking random memory.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_video_template_caps):
- Add support for Indeo-3 (IV32).
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: fix compilation on windows.
+ Remove unused variable when building for windows.
+ Fixes #574443.
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- rewrite
+ * common:
+ Automatic update of common submodule
+ From ffa738d to 7032163
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/video/video.c: (gst_video_get_size):
- * gst/audiotestsrc/gstaudiotestsrc.c:
- doc updates
+ * common:
+ Automatic update of common submodule
+ From 3f13e4e to ffa738d
-2005-10-17 Andy Wingo <wingo@pobox.com>
+2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * common:
+ Automatic update of common submodule
+ From 3c7456b to 3f13e4e
- * gst/tcp/gstmultifdsink.c: Convert to use the boilerplate macro.
+2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst/tcp/gsttcp.c (gst_tcp_socket_read): Comment update.
+ * common:
+ Automatic update of common submodule
+ From 57c83f2 to 3c7456b
- * 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.
+2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c (gst_ogg_mux_send_headers): Fix a segfault
- in strange circumstance.
+ * 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.
-2005-10-17 Julien MOUTTE <julien@moutte.net>
+2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_get_unit_size): We are asked to compute a buffer size
- from caps, let's use the caps...
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add theora mapping
+ Add theora mappings. See #574169.
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * configure.ac:
- put back AX_CREATE_STDINT_H, ffmpegcolorspace includes _stdint.h
+ * 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()
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst-libs/gst/interfaces/Makefile.am:
- fix silly typo
+ * Makefile.am:
+ * win32/common/audio-enumtypes.c:
+ win32: indent copied *-enumtypes.c files in make win32-update
-2005-10-16 Andy Wingo <wingo@pobox.com>
+2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/playback/gstdecodebin.c
- (gst_element_set_state_like_a_crazy_man): New kraaaaaaazy
- function!
- (try_to_link_1): Increase kraziness level.
+ * win32/MANIFEST:
+ win32: update MANIFEST
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* 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
+ * win32/common/config.h:
+ win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
-2005-10-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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).
-2005-10-13 Stefan Kost <ensonic@users.sf.net>
+2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * examples/stats/mp2ogg.c:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- typo fixes
+ * .gitignore:
+ gitignore: ignore more
-2005-10-13 Michael Smith <msmith@fluendo.com>
+2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
- * ext/ogg/gstoggmux.c:
- Use magic glib macros to define constants as 64 bit, to ensure
- appropriate vararg passing.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fix build on Mac OS X
-2005-10-13 Michael Smith <msmith@fluendo.com>
+2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
- * 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.
+ * 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.
-2005-10-12 Stefan Kost <ensonic@users.sf.net>
+2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ rtsp: fix parsing of 'now-' ranges.
+ --
+
+2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
+2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-12 Wim Taymans <wim@fluendo.com>
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ docs: add appsrc symbols to standard section
+ --
- * examples/seeking/Makefile.am:
- Oops.
+2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-12 Wim Taymans <wim@fluendo.com>
+ * gst/adder/gstadder.c:
+ adder: add variants for unsigned to fix warnings for unneeded check
+ For unsigned int out+in can't be < 0.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read), (gst_ring_buffer_clear):
- Don't assert on normal stuff.
+2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstplaybin.c: (do_playbin_seek):
- API fix.
+ * gst/subparse/gstsubparse.c:
+ subparse: use the right variable in debug log, encoding is not yet initialized
-2005-10-12 Wim Taymans <wim@fluendo.com>
+2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * sys/v4l/v4l_calls.c:
+ v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- Cleanups.
- Commit and read from ringbuffer in samples rather than bytes.
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: add missing break in event handling, remove dead code
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Respect segment rate and accum when scheduling samples.
+ * gst-libs/gst/rtsp/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.
-2005-10-11 Julien MOUTTE <julien@moutte.net>
+2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_collected): Quick hack to fix build. We need to handle
- EOS correctly, that needs more work.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * 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.
-2005-10-11 Wim Taymans <wim@fluendo.com>
+2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/generic/states.c: (GST_START_TEST):
- remove old property.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * win32/common/libgstrtsp.def:
+ RTSP: Add support for server tunneling
+ Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
+ that a server can store and match the id against other tunnel requests.
+ Fix the URI in the tunnel requests so that they contain the absolute uri and the
+ query string if any instead of just the hostname.
+ Transparently base64 decode the input stream when tunneling.
+ Add method to set the connection ip address so that it can be included in the
+ tunnel response.
+ Add method to connect the two tunnel requests.
+ Add two callbacks for the async mode to notify a tunnel start and tunnel
+ complete event.
+ Add method to reset the watch after the connection has been tunneled.
+ Various little refactoring to make more stuff reusable.
+ API: RTSP::gst_rtsp_connection_set_ip()
+ API: RTSP::gst_rtsp_connection_get_tunnelid()
+ API: RTSP::gst_rtsp_connection_do_tunnel()
+ API: RTSP::gst_rtsp_watch_reset()
+
+2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_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-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: add new defines for tunneling
+ Add two more result codes for tunneling support.
-2005-10-11 Michael Smith <msmith@fluendo.com>
+2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: remove , from last enum member
+ Remove , from last enum member to improve compatibility with other compilers.
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
- (do_playbin_seek), (gst_play_bin_send_event):
- Override send_event differently, so that we can takes bits of
- functionality from GstPipeline (special handling for seeks,
- including pausing/resuming, and resetting stream time) and still get
- the appropriate behaviour of only forwarding event to a single sink,
- rather than all of them.
- Unfortunately requires a lot of code duplication, but the
- alternatives are equally ugly in the end.
+2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
-2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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>
- * check/elements/audioconvert.c: (setup_audioconvert),
- (cleanup_audioconvert), (get_int_caps), (verify_convert),
- (GST_START_TEST), (audioconvert_suite):
- clean up tests a little, fix some leaks.
+2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: remove debugging g_message
+ --
- * ext/alsa/gstalsasink.c:
- Also allow unsigned int.
+2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- Small cleanup
+ * 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()
-2005-10-10 Wim Taymans <wim@fluendo.com>
+2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Small update, use API as stated in design docs.
+ * 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.
- * examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
- (update_scale), (do_seek), (seek_cb), (set_update_scale),
- (start_seek), (stop_seek), (play_cb), (pause_cb), (stop_cb),
- (message_received), (main):
- Updated seek example for GOption. Some usability improvements.
+2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * win32/common/libgstrtsp.def:
+ win32: Add gst_rtsp_connection_get_url definition
+ No, I'm not wim's buildslave, seriously.
- * gst/audioconvert/audioconvert.h:
- * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
- (gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
- Alloc temp storage somewhere else where we can do it more
- portable.
+2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-10 Wim Taymans <wim@fluendo.com>
+ * 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/tcp/gsttcpserversrc.c: (gst_tcpserversrc_create),
- (gst_tcpserversrc_start):
- Don't block in accept while doing the state change, move
- to poll and make cancellable.
+2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-10-09 Philippe Khalaf <burger@speedy.org>
+ * 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/rtp/rtpbasedepayload.c:
- Set timestamp and add queue delay to timestamp
- * gst-libs/gst/rtp/rtpbuffer.h:
- Set correct payload type for h263
+2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+ * configure.ac:
+ Bump glib requirement to 2.14
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_class_init),
- (gst_audiotestsrc_init), (gst_audiotestsrc_create_sine),
- (gst_audiotestsrc_create_square), (gst_audiotestsrc_create_saw),
- (gst_audiotestsrc_create_triangle),
- (gst_audiotestsrc_create_silence),
- (gst_audiotestsrc_create_white_noise),
- (gst_audiotestsrc_init_pink_noise),
- (gst_audiotestsrc_generate_pink_noise_value),
- (gst_audiotestsrc_create_pink_noise),
- (gst_audiotestsrc_change_wave):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- fixed typo, added pink noise
+2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * ext/gio/gstgiobasesink.c:
+ gio: Use correct format modifier for size_t
+ Fixes bug #573528.
- * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
- (plugin_init):
- Add wavpack and spc typefind functions from 0.8 branch.
+2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Use correct types for some functions on Win32
+ Fixes bug #573529.
- * gst/typefind/gsttypefindfunctions.c: (tar_type_find),
- (ar_type_find), (msdos_type_find), (plugin_init):
- Add typefind functions for tar archives, ar archives,
- RAR archives, and msdos-executables (dlls, exe, etc.).
- Some of those would be wrongly identified as mpeg
- streams of some sort before (#315550).
+2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-09 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix warning about using unitialized value.
- * configure.ac:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type):
- * gst/audiotestsrc/Makefile.am:
- * gst/audiotestsrc/gstaudiotestsrc.c:
- (gst_audiostestsrc_wave_get_type), (gst_audiotestsrc_base_init),
- (gst_audiotestsrc_class_init), (gst_audiotestsrc_init),
- (gst_audiotestsrc_src_fixate), (gst_audiotestsrc_setcaps),
- (gst_audiotestsrc_get_query_types), (gst_audiotestsrc_src_query),
- (gst_audiotestsrc_wait), (gst_audiotestsrc_unlock),
- (gst_audiotestsrc_create_sine), (gst_audiotestsrc_create_square),
- (gst_audiotestsrc_create_saw), (gst_audiotestsrc_create_triangle),
- (gst_audiotestsrc_create_silence),
- (gst_audiotestsrc_create_white_noise),
- (gst_audiotestsrc_change_wave), (gst_audiotestsrc_create),
- (gst_audiotestsrc_set_property), (gst_audiotestsrc_get_property),
- (gst_audiotestsrc_start), (plugin_init):
- * gst/audiotestsrc/gstaudiotestsrc.h:
- add new plugin and element
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init):
- use gobject_class
+2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-09 Tim-Philipp Müller <tim at centricular dot net>
+ * 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
- * gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),
- (gst_adder_init), (gst_adder_request_new_pad),
- (gst_adder_change_state):
- Add query function to source pad, so adder reports the correct
- time/sample position when queried (#315457); fix state change
- function; use GST_DEBUG_FUNCPTR() for pad functions.
+2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-08 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/alsa/gstalsa.c:
+ alsa: release pcminfo after the strdup
- * gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
- Fix leaks in typefind registration
- Clean up the gratuitous commenting and whitespacing a little
+2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtsprange.c:
+ rtsprange: don't leak the range in case of parsing error.
+ Free the gstRTSPTimeRange if we don't return it. Also simplify
+ gst_rtsp_range_free() as it is valid to pass NULL to g_free().
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
- Only actually wait for the thread to be stopped if it's
- running.
+2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * ext/alsa/gstalsa.c:
+ alsa: cleanup name lookup.
+ We can break, once we have a name to make sure, we won't read it ever twice.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render):
- If we receive EOS we can start playback of what we had.
+2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-08 Wim Taymans <wim@fluendo.com>
+ * gst/subparse/gstsubparse.c:
+ subparse: don't leak line, if flushing
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_finalize), (multifdsink_hash_remove),
- (gst_multifdsink_stop):
- Fix crasher when going to NULL multiple times.
+2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ * ext/gio/gstgiosink.c:
+ giosink: reflow error handling to not leak uri
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
- (gst_base_audio_src_create), (gst_base_audio_src_change_state):
- * gst-libs/gst/audio/gstbaseaudiosrc.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
- patch from Edgard Lima <edgard.lima@indt.org.br>
- Fixed gstbaseaudiosrc adding ring buffer sync to it.
+2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-10-06 Wim Taymans <wim@fluendo.com>
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: remove unused code/variables
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
- Report the FLOW_RETURN as string in the error message.
+2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
- Don't assert when clearing an unnegotiated buffer.
+ * sys/ximage/ximagesink.c:
+ ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
-2005-10-04 Michael Smith <msmith@fluendo.com>
+2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/playback/gstplaybasebin.c: (group_destroy),
- (gen_preroll_element), (remove_groups), (setup_source):
- * gst/playback/gstplaybin.c: (remove_sinks), (add_sink),
- (setup_sinks), (gst_play_bin_send_event),
- (gst_play_bin_change_state):
- Set state to NULL before removing from bin. Fix refcounting.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ * win32/common/libgstapp.def:
+ app: add callbacks to appsrc, cleanups
+ Add a uri handler to appsink.
+ don't emit signals when we have installed callbacks on appsink.
+ Add callbacks to appsrc to replace the signals.
+ Add property to disable callbacks in appsrc, default to TRUE for backwards
+ compatibility but disable when callbacks are installed.
+ API: GstAppSrc::emit-signals
+ API: GstAppSrc::gst_app_src_set_emit_signals()
+ API: GstAppSrc::gst_app_src_get_emit_signals()
+ API: GstAppSrc::gst_app_src_set_callbacks()
+
+2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-10-04 Michael Smith <msmith@fluendo.com>
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.h:
+ * tests/check/elements/appsink.c:
+ Appsink: add padding for callbacks + docs
+ Add some padding to the callbacks structure just to be safe.
+ Remove the now invisible marshaller methods from the docs.
+ Fix a comment in the unit test.
- * gst/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.
+2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ * win32/common/libgstapp.def:
+ win32: Add new libgstapp symbol
- * configure.ac:
- back to development
+2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
-=== release 0.9.3 ===
+ * 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.
-2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- Releasing 0.9.3, "De Facto"
+ * gst/playback/gstplaybasebin.c:
+ docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
-2005-10-03 Andy Wingo <wingo@pobox.com>
+2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
- * gst/playback/gstdecodebin.c (try_to_link_1)
- (remove_element_chain): set element to NULL before removing it.
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/playback/gstplaybin2.c:
+ docs: playbin2 has no stream-info
-2005-10-02 Johan Dahlin <johan@gnome.org>
+2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
- * ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_uri_get_protocols):
- protect gst_gnomevfs_get_supported_uris by a mutex, to make it
- MT safe.
+ * gst-libs/gst/video/video.h:
+ docs: fix newly added interlace constants and plug holes in video format docs
-2005-10-02 Andy Wingo <wingo@pobox.com>
+2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ docs: don't put random stuff in tags.
+ Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
+ tag to append text again to the documentation body.
-2005-09-29 Wim Taymans <wim@fluendo.com>
+2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * sys/ximage/ximagesink.c:
+ ximagsink: do not access uninitialized height variable.
+ Exit like in xvimagesink, if we have partial caps.
-2005-09-29 Wim Taymans <wim@fluendo.com>
+2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- We use fixed caps.
+ * Makefile.am:
+ * configure.ac:
+ * win32/common/config.h.in:
+ Change how win32/common/config.h is updated
+ Generate win32/common/config.h-new directly from config.h.in,
+ using shell variables in configure and some hard-coded information.
+ Change top-level makefile so that 'make win32-update' copies the
+ generated file to win32/common/config.h, which we keep in source
+ control. It's kept in source control so that the git tree is
+ buildable from VS.
+ This change is similar to the one recently applied to GStreamer,
+ except that it adds a few -base specific defines.
+
+2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * win32/common/libgstapp.def:
+ app: add win32 .def file and only export functions we want exported
+ Add a .def file for win32 builds (and make check-exports).
+ Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
+ Make sure private marshaller functions aren't exported by prefixing them with __gst;
+ also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
+ a comment why we're not using glib-genmarshal for this one.
+
+2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/dynamic/.gitignore:
+ * tests/examples/dynamic/Makefile.am:
+ * tests/examples/dynamic/sprinkle.c:
+ sprinkle: Add another example app
+ Add an example app that dynamically adds and removes audiotestsrc elements from
+ adder.
- * gst/playback/Makefile.am:
- * gst/playback/test5.c: (new_pad), (no_more_pads), (start_finding),
- (dump_element_stats), (main):
- Added example stream introspection code.
+2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
-2005-09-28 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fixed a typo.
- * gst/adder/gstadder.c: (gst_adder_collected):
- fix adder for float elements
+2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
-2005-09-28 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst/tcp/gstmultifdsink.c:
+ rtsp, multifdsink: Unify the use of union gst_sockaddr.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_class_init),
- (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_class_init),
- (gst_base_audio_src_provide_clock):
- get_clock -> provide_clock
+2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-09-28 Andy Wingo <wingo@pobox.com>
+ * common:
+ * configure.ac:
+ build: Update shave init statement for changes in common. Bump common.
- * gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
- and unlocking.
+2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/gstsinesrc.c (gst_sinesrc_unlock): Actually implement
- unlocking.
+ * 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.
- * gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
- Actually add the pad template.
- (gst_tcpclientsink_get_type): We're a base sink. Woot, works.
+2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
- I'm at it...
+ * 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.
- * gst/tcp/gsttcpclientsrc.c: Make interruptable -- code stolen
- from fdsrc. Get caps in create() instead of start() so it can be
- interrupted. Interruption somewhat untested.
+2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
- Proper EOS handling.
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ playsink: Add FLUSHING pad type
+ Make it possible to request a flushing pad from the playsink. We can eventually
+ use these flushing pads to quickly terminate the dataflow when we are shutting
+ down.
-2005-09-27 Andy Wingo <wingo@pobox.com>
+2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/tcp/gsttcpclientsrc.c: Cleaned up.
+ * common:
+ Automatic update of common submodule
+ From 9cf8c9b to a6ce5c6
- * gst/tcp/gsttcpserversrc.c: Cleaned up.
+2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
+ Fixes: #565777
- * gst/tcp/gsttcp.h:
- * gst/tcp/gsttcp.c (gst_tcp_read_buffer): New function, factored
- out of tcpclientsrc.c. Cancellable.
- (gst_tcp_socket_read): Made private, cancellable, with better
- diagnostics. Also the FIONREAD ioctl takes a int*, not a size_t*.
- (gst_tcp_gdp_read_buffer): Made cancellable, actually returns the
- whole buffer, and better diagnostics.
- (gst_tcp_gdp_read_caps): Same.
+2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
+ * tests/icles/stress-playbin.c:
+ stress-playbin: print the current uri
+ Print the current uri so that we can more easily see what uri caused a crash or
+ error.
-2005-09-26 Andy Wingo <wingo@pobox.com>
+2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * tests/icles/stress-playbin.c:
+ Print the errors more clearly
+ Print some more verbose messages when dealing with errors.
- * gst/volume/gstvolume.c (volume_transform_ip): Update for
- controller api change.
+2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-24 Wim Taymans <wim@fluendo.com>
+ * 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.
- * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
- (gst_audioringbuffer_stop):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render),
- (gst_base_audio_sink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
- (gst_ring_buffer_commit), (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- Fix sync again. Moved sample alignment to basesink.
+2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * docs/plugins/Makefile.am:
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * gst/volume/gstvolume.c:
- add/fix docs
- * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size):
- * gst-libs/gst/audio/audio.h:
- add conversion macros for frames <-> clocktime
+2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-23 David Schleef <ds@schleef.org>
+ * gst/playback/gstplaybin2.c:
+ Playbin2: Don't leave unused parameters in debug statements.
+ Fixes build on macosx
- * gst/audioresample/Makefile.am:
- * gst/audioresample/debug.h:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/resample.c: Convert to using gst debugging
+2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c:
+ Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_send_event):
- Only seek on one sink, the first one that succeeds.
+2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-22 Michael Smith <msmith@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Add some G_UNLIKELY because we can
+ Add a G_UNLIKELY when checking the shutdown variable.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_change_state):
- Don't flush encoder state unless we have an initialised encoder.
- Clear out encoder state on PAUSED_TO_READY.
+2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
-2005-09-22 Wim Taymans <wim@fluendo.com>
+ * 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
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_is_filled), (gst_basertppayload_push),
- (gst_basertppayload_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Added max-ptime to control amount of data in the rtp packets.
+2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-09-21 Andy Wingo <wingo@pobox.com>
+ * gst/tcp/gstmultifdsink.c:
+ multifdsink: Fix strict aliasing error using a union
- * gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
- thingies.
+2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
- can be called multiple times, dogs.
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Fix a strict aliasing warning
+ Fix strict aliasing warnings from casting a sockaddr_storage and
+ using it as a sockaddr_in6. Use a union instead.
-2005-09-21 Wim Taymans <wim@fluendo.com>
+2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
- * gst-libs/gst/rtp/gstbasertppayload.c:
- (gst_basertppayload_class_init), (gst_basertppayload_init),
- (gst_basertppayload_setcaps), (gst_basertppayload_chain),
- (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
- (gst_basertppayload_push), (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- Allow 0 ssrc too.
+ * docs/libs/.gitignore:
+ * docs/libs/tmpl/.gitignore:
+ * docs/plugins/.gitignore:
+ * docs/plugins/tmpl/.gitignore:
+ Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
-2005-09-21 Stefan Kost <ensonic@users.sf.net>
+2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * docs/libs/compiling.sgml:
- fixing typos
+ * docs/plugins/Makefile.am:
+ * ext/vorbis/Makefile.am:
+ * ext/vorbis/gstvorbisdec.h:
+ * ext/vorbis/gstvorbisenc.h:
+ * ext/vorbis/gstvorbisparse.h:
+ * ext/vorbis/gstvorbistag.h:
+ * ext/vorbis/vorbis.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisenc.h:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbisparse.h:
+ * ext/vorbis/vorbistag.c:
+ * ext/vorbis/vorbistag.h:
+ vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
-2005-09-20 Wim Taymans <wim@fluendo.com>
+2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * 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.
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add conversion from/to YVYU colorspace
+ Fixes bug #572872.
-2005-09-20 Wim Taymans <wim@fluendo.com>
+2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.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_set_property),
- (gst_basertppayload_get_property),
- (gst_basertppayload_change_state):
- * gst-libs/gst/rtp/gstbasertppayload.h:
- Make timestamp offset configurable.
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
+ The conversion from UYVY to RGB24 and then to GRAY8
+ is quite slow. Fixes bug #569655.
-2005-09-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- fix wrong pop/unref
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix deadlock when shutting down. Fixes #572577.
-2005-09-19 Torsten Schoenfeld <kaffeetisch at gmx dot net>
+2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * tests/icles/stress-playbin.c:
+ stress-playbin: make more flexible, e.g. also useful for playbin2
- * 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)
+2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * 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.
+2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-09-19 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaysink.c:
+ playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
- * examples/seeking/seek.c: (make_wav_pipeline), (main):
- Fixed wav pipeline.
+2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
-2005-09-19 Wim Taymans <wim@fluendo.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-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>
- * 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.
+2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-09-19 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Use shave for the build output
- * 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.
+2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-19 Wim Taymans <wim@fluendo.com>
+ * win32/common/libgstrtsp.def:
+ win32: Add new symbol to libgstrtsp.def
- * check/pipelines/simple_launch_lines.c: (run_pipeline):
- Fix for bus API.
+2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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-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.
+2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-09-18 David Schleef <ds@schleef.org>
+ * tests/check/libs/netbuffer.c:
+ Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
- * 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)
+2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-09-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/theora/theoraparse.c:
+ theoraparse: Use the correct unref functions
- * check/elements/audioconvert.c: (GST_START_TEST):
- try out 24 bit conversion
+2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
- * 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.
+2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-libs/gst/tag/gsttagdemux.c:
+ tagdemux: Unref the actual buffer instead of the memory address of the buffer
- * 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
+2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ Automatic update of common submodule
+ From 5d7c9cc to 9cf8c9b
- * 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
+2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgstvideo.def:
+ win32/common: Update .def files for recent API addition
- * common/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 :)
+2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-15 Wim Taymans <wim@fluendo.com>
+ * tests/check/libs/rtp.c:
+ tests: Fix indentation
- * gst/audioconvert/gstaudioconvert.c:
- And enable 24 bits mode as well..
+2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-09-15 Wim Taymans <wim@fluendo.com>
+ * gst-libs/gst/video/video.c:
+ libs/video: Fix gst_video_format_new_caps* functions.
+ Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
+ don't add anything.
- * gst-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.
+2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
-2005-09-15 Andy Wingo <wingo@pobox.com>
+ * common:
+ Automatic update of common submodule
+ From 80c627d to 5d7c9cc
- * configure.ac (plugindir): Remove the EOL matcher from the
- regexp, as it causes me problems. Libtool? Make? Who knows?
+2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-14 David Schleef <ds@schleef.org>
+ * 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.
- * 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.
+2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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.
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * 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.
-2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
- add theoraenc
+ theoraenc: Add property for speed level control
+ Add property "speed-level" to control the amount of motion searching
+ the encoder does. This is only available in libtheora >= 1.0 and
+ will silently fail with earlier libraries. Fixes: #572275.
+ Signed-off-by: David Schleef <ds@schleef.org>
-2005-09-13 Tim-Philipp Müller <tim at centricular dot net>
+2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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-libs/gst/video/video.c:
+ * gst-libs/gst/video/video.h:
+ video: Fix 'Since' tags
+
+2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/test4.c: (main):
- Fix MinGW build problem and use g_usleep() instead of
- sleep() (#316162)
+ * 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).
-2005-09-12 Wim Taymans <wim@fluendo.com>
+2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
-2005-09-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
- * docs/plugins/gst-plugins-base-plugins-docs.sgml:
- * docs/plugins/gst-plugins-base-plugins-sections.txt:
- * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
- * docs/plugins/tmpl/element-tcpserversink.sgml:
- * gst/ffmpegcolorspace/gstffmpeg.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/volume/gstvolume.c:
- add more elements to the docs
-
-2005-09-09 Jan Schmidt <thaytan@mad.scientist.com>
-
- * check/Makefile.am:
- * check/pipelines/simple_launch_lines.c: (setup_pipeline),
- (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
- Add extra tests for basetransform based components.
- Comment out the test_element_negotiation test until we decide
- if it's testing correct behaviour.
- * ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
- (gst_visual_chain), (gst_visual_change_state):
- Slightly more correct but still bogus timestamping.
- Fix state change function.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init):
- * gst/audioresample/gstaudioresample.c:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_class_init):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_prepare_size), (gst_videoscale_set_caps),
- (gst_videoscale_prepare_image):
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform_ip):
- Basetransform updates. Enable passthrough modes.
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
- (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
- Negotiation fix that allows the window to return to the original
- size and renegotiate passthrough upstream. Extra debug output.
-
-2005-09-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/sine/gstsinesrc.c:
- * gst/volume/gstvolume.c:
- fix up header include
+ * gst-libs/gst/riff/riff-media.c:
+ Add some more mappings for h264 in riff
-2005-09-09 Stefan Kost <ensonic@users.sf.net>
+2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * win32/common/libgstrtsp.def:
+ Add new RTSP symbols to def files
+ Add the new RTSP symbols to the windows def file.
-2005-09-08 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
- (gst_xvimagesink_check_xshm_calls):
- Fix compilation when XShm is not available.
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/appsink.c:
+ Add method to install callbacks on appsink
+ Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
+ Fixes #571299.
+ Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
+ performant alternative to connecting to the signals.
+ Add a unit test for appsink.
+ Clean up some of the appsink docs.
+ API: GstAppSink::gst_app_sink_set_callbacks()
+
+2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-07 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:
+ 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/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.
+2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+ * 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>
- * ext/vorbis/vorbisenc.c:
- * gst-libs/gst/tag/gstvorbistag.c:
- gsttaginterface.h -> gsttagsetter.h
+ * 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.
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * check/pipelines/simple_launch_lines.c: (GST_START_TEST):
- added another test that failes for me (test is not active by default)
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/videotestsrc/videotestsrc.c:
+ ffmpegcolorspace/videotestsrc: Use v308 instead of V308
-2005-09-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- v4l2 is no longer in gst-plugins-base
+ * 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.
-2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * configure.ac:
- In the output at the end, don't show the first plugin on the same
- line as "Core plug-ins, always built:".
- Indent the output as for other plugin categories
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_create):
- #define that can be used to not use peer buffer_alloc functions for
- test purposes.
- * sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
- (gst_ximage_buffer_get_type), (gst_ximagesink_ximage_new),
- (gst_ximagesink_show_frame):
- * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_init),
- (gst_xvimage_buffer_get_type), (gst_xvimagesink_setcaps),
- (gst_xvimagesink_show_frame):
- Error case handling fixes. gst-launch fakesrc ! x[v]imagesink now
- fails gracefully instead of XError aborting or deadlocking.
-
-2005-09-07 Stefan Kost <ensonic@users.sf.net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
+ Partially fixes bug #571147.
- * ext/libvisual/Makefile.am:
- link against gst-base-libs
+2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-2005-09-06 David Schleef <ds@schleef.org>
+ * 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).
- * configure.ac: Enable libvisual plugin.
- * ext/libvisual/Makefile.am:
- * ext/libvisual/visual.c: Fixes to make it compile.
+2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-=== release 0.9.2 ===
+ * 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.
-2005-09-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * NEWS:
- * RELEASE:
- * configure.ac:
- * docs/random/ChangeLog-0.8:
- releasing 0.9.2, "Spoon"
+ * .gitignore:
+ Make git ignore backup files.
-2005-09-05 Michael Smith <msmith@fluendo.com>
+2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
- * 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.
+ * 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.
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
- * common/gtk-doc-plugins.mak:
- * docs/plugins/Makefile.am:
- fix distcheck
- * gst/audioresample/resample.c:
- fix wrong docstring
+ * 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.
-2005-09-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
- * common/gst-xmlinspect.py:
- * common/gtk-doc-plugins.mak:
- only inspect plugins for this given package
- require gst-python 0.9
+ * 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.
-2005-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
- * 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
+ Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
-2005-08-31 Wim Taymans <wim@fluendo.com>
+2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Resync if the buffer timestamps drift more than a 10th
- of a second.
-
-2005-08-31 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
+ Partially fixes bug #571147.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_property),
- (gst_v4lsrc_get_property):
- The 'timestamp-offset' property is registered as an int64, so
- let's use g_value_{set|get}_int64() in our setter and getter
- functions (makes it work and fixes warnings with gst-inspect).
+2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
-2005-08-30 Wim Taymans <wim@fluendo.com>
+ * 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().
- * check/elements/audioconvert.c: (setup_audioconvert):
- * check/elements/audioresample.c: (setup_audioresample):
- * check/elements/volume.c: (setup_volume):
- Fix checks.
+2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
-2005-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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>
- * common/gtk-doc-plugins.mak:
- * common/plugins.xsl:
- * docs/plugins/Makefile.am:
- make module a param
+ * 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.
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * 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()
-2005-08-30 Stefan Kost <ensonic@users.sf.net>
+2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/volume/gstvolume.c: (gst_volume_class_init),
- (volume_transform):
- do not update controlled params, if buffer has no timestamp
+ * 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.
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
- * configure.ac:
- * gst/sine/Makefile.am:
- * gst/volume/Makefile.am:
- controllerized elements also need to link against controller-libs ;)
+ * gst/playback/gstplaysink.c:
+ playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
+ The flags where present but actually not been taken into account.
-2005-08-29 Stefan Kost <ensonic@users.sf.net>
+2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.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
+ * 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.
-2005-08-29 Andy Wingo <wingo@pobox.com>
+2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/vorbis/vorbisdec.c (vorbis_dec_convert, vorbis_dec_push)
- (vorbis_handle_data_packet): Fix some int overflow errors.
+ * 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.
- * 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.
+2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
- * check/elements/volume.c (cleanup_volume): Fix for running
- CK_FORK=no.
+ * 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.
- * check/elements/audioconvert.c: Convert from native endian, not
- little endian.
+2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-29 Michael Smith <msmith@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.
- * 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>
+2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioresample/gstaudioresample.c:
- use base class' newsegment to properly timestamp
+ * win32/common/libgstrtsp.def:
+ Add new methods
+ Add new methods to the windows def file.
-2005-08-26 Wim Taymans <wim@fluendo.com>
+2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
- * 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.
+ * 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.
-2005-08-26 Wim Taymans <wim@fluendo.com>
+2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * 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.
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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.
+ * configure.ac:
+ * gst/audioresample/gstaudioresample.c:
+ Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
-2005-08-26 Jan Schmidt <thaytan@mad.scientist.com>
+2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
- (vorbisdec_finalize), (vorbis_handle_type_packet):
- Big shout-out to MikeS for fixing this giant memory leak.
- Huzzah!
+ * sys/ximage/ximagesink.c:
+ Fix buffer_alloc in ximagesink
+ Remove some useless debug info that reported wrong image sizes.
+ When upstream does not accept out suggested size, fall back to allocating an
+ image of the requested width/height instead of the currently configured size.
+ The problem is that an image is reused from the pool because the width/height
+ match but the caps on the new buffer are the requested caps with possibly
+ different height/width resulting in errors.
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose),
- (audio_convert_get_unit_size):
- plug some leaks
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gsturidecodebin.c:
+ Fix documentation for autoplug-select
+ fix the documentation strings for the autoplug-select signal.
+ Fixes #570142.
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * check/Makefile.am:
- * check/elements/audioconvert.c: (setup_audioconvert),
- (cleanup_audioconvert), (get_int_caps), (verify_convert),
- (GST_START_TEST), (audioconvert_suite), (main):
- add a test for audioconvert
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
- note that for buffers of 1/3 sec this means DURATION(c) is
- one nanosecond more than for a and b
-
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check/Makefile.am:
- * check/elements/audioresample.c: (setup_audioresample),
- (cleanup_audioresample), (fail_unless_perfect_stream),
- (test_perfect_stream_instance), (GST_START_TEST),
- add a check for audioresample
- (audioresample_suite), (main):
- * check/elements/volume.c: (GST_START_TEST):
- remove unused method
- * gst/audioresample/gstaudioresample.c:
- set correct buffer parameters since we're changing them
- * gst/audioresample/resample_ref.c: (resample_scale_ref):
- add some debug
+ * 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.
-2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
- * 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
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ Its "Base Library" and not just "Library".
-2005-08-25 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- Revert unpopular change for GST_MESSAGE_SRC to GObject.
+2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-25 Stefan Kost <ensonic@users.sf.net>
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ Link to the class, as we can't link to the members yet.
- * gst/volume/gstvolume.c:
- made set_caps function static
+2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Wim Taymans <wim@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.
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_change_state):
- Stop leaking taglists.
+2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Unref selector request pad even if we no longer have a selector.
+ During destruction, we won't have a selector any more, but we still need
+ to unref the pad to avoid leaking it.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
- (gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
- (gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
- (gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
- Parse seeking events better.
- Unref static caps.
- Generate correct newsegment events, fixes seeking in live oggs.
+2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
- * ext/theora/theoradec.c: (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_src_getcaps),
- (theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
- Use newsegment values to report correct play time.
+ * gst/playback/gstplaybin2.c:
+ Unref source in playbin2's finalize method
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_sink_event):
- * ext/vorbis/vorbisdec.h:
- Parse and use newsegment values to report correct play time.
+2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
- * 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.
+ * 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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
- Basesink does newsegment calculations for us now.
+2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 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.
- * check/Makefile.am:
- * configure.ac:
- add core's plugins to the mix so that playbin works
- * check/generic/states.c: (GST_START_TEST):
- set a 0 timeout on pipelines, so they don't force the next
- state change
- * gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
- (gst_play_base_bin_change_state):
- remove the crappy error handling and do GST error handling
+2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ Unref peer request pad after releasing it, since we hold a reference.
- * check/Makefile.am:
- * check/generic/states.c: (GST_START_TEST), (states_suite), (main):
- add same test as to core, it bitches out on playbin atm.
+2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Fix caps leak in playbin2.
- * configure.ac:
- Remove audioscale.
+2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Wim Taymans <wim@fluendo.com>
+ * gst/playback/gstplaybin2.c:
+ Unref active pad from selector when finding active stream.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
- (gst_videoscale_prepare_size), (parse_caps),
- (gst_videoscale_set_caps), (gst_videoscale_get_size),
- (gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
- (gst_videoscale_transform):
- * gst/videoscale/gstvideoscale.h:
- Refactor, make use of BaseTranform really well.
+2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gstplaybin2.c:
+ Free uris when finalizing playbin2 instance.
- * configure.ac:
- compile audioresample
- * gst/audioresample/Makefile.am:
- * gst/audioresample/buffer.c:
- * gst/audioresample/functable.c:
- * gst/audioresample/gstaudioresample.c:
- * gst/audioresample/gstaudioresample.h:
- * gst/audioresample/resample.c:
- (resample_get_output_size_for_input):
- * gst/audioresample/resample.h:
- * gst/audioresample/resample_chunk.c:
- * gst/audioresample/resample_functable.c:
- * gst/audioresample/resample_ref.c:
- port to use basetransform; doesn't work in all cases yet
+2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
-2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/playback/gsturidecodebin.c:
+ Unref pads when iterating over them in analyse_source.
+ Fixes leak of source's srcpad when using uridecodebin.
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_init),
- (audio_convert_get_unit_size), (audio_convert_transform_caps),
- (audio_convert_fixate_caps), (audio_convert_set_caps),
- (audio_convert_transform),
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format),
- (gst_audio_convert_channels):
- * gst/audioconvert/gstchannelmix.c:
- * gst/audioconvert/gstchannelmix.h:
- port to basetransform
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_get_unit_size):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
- (gst_videoscale_transform_caps), (gst_videoscale_get_unit_size):
- fix for basetransform changes
+2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-24 Jan Schmidt <thaytan@mad.scientist.com>
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ Add releaseinfo with online url.
- * check/Makefile.am:
- Add CHECK_CFLAGS and LDFLAGS
+2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
- * gst/playback/gstplaybasebin.c: (fill_buffer):
- GST_MESSAGE_SRC became a GObject
+ * gst/playback/gstplaybasebin.c:
+ Fix compilation warning on Forte
-2005-08-24 Wim Taymans <wim@fluendo.com>
+2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
- * 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/adder/gstadder.c:
+ Don't do void pointer arithmetic.
-2005-08-24 Andy Wingo <wingo@pobox.com>
+2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
- * sys/v4l/gstv4lelement.c (gst_v4lelement_start)
- (gst_v4lelement_stop): Call _start and _stop for xoverlay instead
- of _open and _close.
+ * common:
+ Bump common
- * 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:
+2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-23 Wim Taymans <wim@fluendo.com>
+ * autogen.sh:
+ * common:
+ Use a symbolic link for the pre-commit client-side hook
- * examples/seeking/seek.c: (make_vorbis_pipeline),
- (make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
- Small seek updates.
+2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-23 Andy Wingo <wingo@pobox.com>
+ * .gitignore:
+ Add more files/directories to ignore
- * gst-libs/gst/audio/gstbaseaudiosrc.c
- (gst_base_audio_src_fixate): Only fixate endianness if it is
- present in the caps.
+2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-22 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ fix some typos
+ Fix some typos in the doc string of the new
+ gst_rtsp_options_as_string() method.
- * ext/alsa/gstalsasink.c (gst_alsasink_get_property):
- * ext/alsa/gstalsasrc.c (gst_alsasrc_get_property): Add a
- device-name property.
+2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstaudiosrc.h:
- * gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
- close_device in the ring buffer, like gstaudiosink.
+ * 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/alsa/gstalsamixer.h:
- * ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
- macro to implement the interface without much code. Cleanups.
+2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Add new method to docs
+ Add the new gst_rtsp_options_as_text() method to the docs.
-2005-08-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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/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()
- * 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.
+2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
- * 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/typefind/gsttypefindfunctions.c:
+ Ensure we have sufficient data when using data scan contexts.
+ Fixes crashes typefinding things that look like they might contain AAC
+ data (but probably aren't actually AAC).
- * gst-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.
+2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-18 Stefan Kost <ensonic@users.sf.net>
+ * ext/gio/Makefile.am:
+ Fix include order for gio plugin
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_query):
- moved statement below switch
- * gst/volume/gstvolume.c: (gst_volume_class_init):
- added debug ptr
+2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-16 Wim Taymans <wim@fluendo.com>
+ * win32/common/config.h:
+ Update win32 config.h for 0.10.22.1 dev cycle
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_change_state):
- Open and close device in READY<->NULL state change.
+2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-16 Andy Wingo <wingo@pobox.com>
+ * .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
- * examples/seeking/Makefile.am: Don't compile non-compiling
- compiled objects with the compiler.
+2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * examples/seeking/seek.c (make_dv_pipeline): Update for new DV
- elements.
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/README:
+ * gst/audioresample/arch.h:
+ * gst/audioresample/buffer.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/debug.c:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/fixed_arm4.h:
+ * gst/audioresample/fixed_arm5e.h:
+ * gst/audioresample/fixed_bfin.h:
+ * gst/audioresample/fixed_debug.h:
+ * gst/audioresample/fixed_generic.h:
+ * gst/audioresample/functable.c:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample.h:
+ * gst/audioresample/resample_chunk.c:
+ * gst/audioresample/resample_functable.c:
+ * gst/audioresample/resample_ref.c:
+ * gst/audioresample/resample_sse.h:
+ * gst/audioresample/speex_resampler.h:
+ * gst/audioresample/speex_resampler_double.c:
+ * gst/audioresample/speex_resampler_float.c:
+ * gst/audioresample/speex_resampler_int.c:
+ * gst/audioresample/speex_resampler_wrapper.h:
+ * gst/speexresample/Makefile.am:
+ * gst/speexresample/README:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/fixed_arm4.h:
+ * gst/speexresample/fixed_arm5e.h:
+ * gst/speexresample/fixed_bfin.h:
+ * gst/speexresample/fixed_debug.h:
+ * gst/speexresample/fixed_generic.h:
+ * gst/speexresample/gstspeexresample.c:
+ * gst/speexresample/gstspeexresample.h:
+ * gst/speexresample/resample.c:
+ * gst/speexresample/resample_sse.h:
+ * gst/speexresample/speex_resampler.h:
+ * gst/speexresample/speex_resampler_double.c:
+ * gst/speexresample/speex_resampler_float.c:
+ * gst/speexresample/speex_resampler_int.c:
+ * gst/speexresample/speex_resampler_wrapper.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/audioresample.c:
+ * tests/check/elements/speexresample.c:
+ Rename files and types from speexresample to audioresample
+ Rename files and types from speexresample to audioresample
+ to finish the move and to prevent any confusion.
-2005-08-12 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.
+2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-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.
+ * sys/xvimage/xvimagesink.c:
+ Add some more debugging to the Xv strides
+ Add some more debugging to the strides as they are received from the server and
+ the expected strides.
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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!)
+ * gst/typefind/gsttypefindfunctions.c:
+ Add typefind function for gsm
+ Because core now supports typefindfactories without a typefind function we can
+ register a factory fo GSM that will --if all else fails-- assume the file is a
+ GSM file based on the registered extension.
+ Fixes #566661.
-2005-08-09 Tim-Philipp Müller <tim at centricular dot net>
+2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * gst/playback/gsturidecodebin.c:
+ Use more performant link function
+ We can use gst_element_link_pads() instead of the more generic
+ gst_element_link() function because we know the pads. This saves some cycles
+ because the more generic function needs to search for possible compatible caps
+ etc.
-2005-08-08 Edward Hervey <edward@fluendo.com>
+2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+ * gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
- backported updates since branch
+ Add more codec ids for RIFF formats
+ Handle codec ID for various other AAC formats.
+ Sync the list of possible codec ids with that of ffmpeg.
+ Fixes #567255
-2005-08-08 Andy Wingo <wingo@pobox.com>
+2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * 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.
- * 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.
+2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * 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.
+ * ext/alsa/gstalsasink.c:
+ Improve debug message
+ Improve the debug message when alsa returns an error.
- * ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
- (gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
- base class API.
+2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst-libs/gst/audio/gstringbuffer.h
- (GstRingBufferClass.open_device, GstRingBufferClass.close_device):
- New vmethods.
+ * 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
- * 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.
+2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
-2005-08-08 Tim-Philipp Müller <tim at centricular dot net>
+ * gst/typefind/gsttypefindfunctions.c:
+ Add typefinder for Mobile XMF. Fixes bug #568707.
- * gst-libs/gst/interfaces/mixer.h:
- Reset padding to GST_PADDING.
+2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ Fix linking on Solaris. Fixes bug #568482.
+ Check for nsl and socket libraries and add them to
+ LIBS if they're found. They're needed for socket()
+ and gethostbyname() on Solaris.
- * gst/playback/gstplaybin.c: (remove_sinks):
- Remove visualization from parent explicitely; works around some
- apparent refcount issue that I haven't tracked down yet.
+2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/playback/gstplaybasebin.c:
+ Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
- * ext/alsa/gstalsasink.c: (set_hwparams):
- Assign debug category, add negotiation debug msgs.
+2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
-2005-08-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * common:
+ Update common snapshot.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_start):
- Fix error code for file-not-found to NOT_FOUND.
+2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ Fix pre-commit hook
- * 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
+2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-2005-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
- * 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
+2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.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):
- Enable videoscale.
+2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
-2005-08-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ Back to devel -> 0.10.22.1
- * gst-libs/gst/gconf/gconf.c:
- * gst-libs/gst/gconf/gconf.h:
- Fix some Andy Problem [tm].
+2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
-2005-08-04 Andy Wingo <wingo@pobox.com>
+ * autogen.sh:
+ * common:
+ Install and use pre-commit indentation hook from common
- * gst/videoscale/gstvideoscale.c (gst_videoscale_get_size):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c
- (gst_ffmpegcsp_get_size): Adapt to API changes.
+2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
- * gst/videoscale/gstvideoscale.c (gst_videoscale_transform_ip):
- Implement an in-place do-nothing transform.
+ * 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.
-2005-08-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ * autogen.sh:
+ autogen.sh : Use git submodule
-2005-08-03 Andy Wingo <wingo@pobox.com>
+=== release 0.10.22 ===
- * sys/ximage/ximagesink.c (gst_ximagesink_renegotiate_size):
- (gst_ximagesink_buffer_alloc):
- Protect the height, width, and desired_caps with the pool_lock.
- Fixes videotestsrc ! queue ! ximagesink.
+2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-08-02 Edward Hervey <edward@fluendo.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * po/LINGUAS:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/config.h:
+ Release 0.10.22
+ Original commit message from CVS:
+ Release 0.10.22
- * gst/volume/gstvolume.c:
- include left from controller cleanup
+2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-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.
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
+
+2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
+ Original commit message from CVS:
+ * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
+ * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
+ * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
+ * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
+ * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
+ * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
+ Use correct struct alignment everywhere to prevent unaligned
+ memory accesses, resulting in SIGBUS on sparc and probably others.
+ Fixes bug #500833.
+
+2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
+ Forward unknown events upstream to allow latency configuration.
+ Fixes bug #567960.
+
+2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (groups_set_locked_state):
+ Provide the right arguments to a debug line.
+
+2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Don't reset the colorkey when element is reused. Fixes #567511.
+
+2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.21.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.21.3 pre-release
+
+2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c:
+ Store the returned signal id in the right slot when
+ registering the pull-buffer signal.
+ Fixes #567168
+ Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
+
+2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/mixer.c:
+ Small docs addition to clarify that one really mustn't free
+ the constant GList returned (#566812).
+
+2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add GType for GstRTSPUrl and expose a copy function because we can.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
+ (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
+ * gst-libs/gst/rtsp/gstrtspurl.h:
+ * win32/common/libgstrtsp.def:
+ Add GType for GstRTSPUrl and expose a copy function because we can.
+ API: gst_rtsp_url_copy()
+ Fixes #567027.
+
+2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add plugin dependency for the GIO and GVfs modules.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gio/gstgio.c: (plugin_init):
+ Add plugin dependency for the GIO and GVfs modules.
+ Fixes bug #566876.
+
+2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add plugin dependency for the gnomevfs modules.
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ Add plugin dependency for the gnomevfs modules.
+ Fixes bug #566875.
+
+2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
+ Original commit message from CVS:
+ * win32/common/libgstcdda.def:
+ Add new symbol to the list of exported symbols.
+
+2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Fix some comments and docs.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
+ (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
+ (activate_group), (deactivate_group), (groups_set_locked_state),
+ (gst_play_bin_change_state):
+ Fix some comments and docs.
+ Post an error message when we fail to link the selector to the sink.
+ Remove pushing of EOS, this seems unneeded.
+ Lock the state of deactivated groups so that they don't accidentally
+ reactivate when the playbin2 state changes.
+ Reuse uridecodebins.
+ Unlock and relock state of groups when playbin goes to NULL.
+ Fixes #566654.
+ Fixes #566341.
+ * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
+ Only do something in the pad removed callback when we are dealing with
+ our sourcepads because the sinkpads don't have a ghostpad.
+
+2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ Make the GType of GstCDDABaseSrcMode public for bindings.
+ Fixes bug #566837.
+
+2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/libvisual/visual.c: (plugin_init):
+ Use new core API to make registry re-scan the plugin
+ whenever visualisations are added or removed (see #350477).
+
+2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
+
+ gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
+ Original commit message from CVS:
+ Patch by: José Alburquerque <jaalburqu svn gnome org>
+ * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
+ * gst-libs/gst/audio/gstaudioclock.h:
+ Make gst_audio_clock_new use const gchar* to ease the wrapping of
+ C++ bindings. Fixes #566723.
+
+2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Add pkg-config files for libgstapp. Fixes bug #566761.
+ Original commit message from CVS:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-app-uninstalled.pc.in:
+ * pkgconfig/gstreamer-app.pc.in:
+ Add pkg-config files for libgstapp. Fixes bug #566761.
+
+2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsink.h:
+ * gst-libs/gst/app/gstappsrc.c:
+ * gst-libs/gst/app/gstappsrc.h:
+ Make debug categories static. Use _element_class_set_details_simple().
+
+2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
+ (gst_app_sink_class_init), (gst_app_sink_init),
+ (gst_app_sink_dispose), (gst_app_sink_finalize),
+ (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
+ (gst_app_sink_flush_unlocked), (gst_app_sink_start),
+ (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
+ (gst_app_sink_render), (gst_app_sink_getcaps),
+ (gst_app_sink_set_caps), (gst_app_sink_get_caps),
+ (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
+ (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
+ (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
+ (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
+ (gst_app_sink_pull_buffer)::
+ * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
+ * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
+ (gst_app_src_class_init), (gst_app_src_init),
+ (gst_app_src_flush_queued), (gst_app_src_dispose),
+ (gst_app_src_finalize), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_unlock),
+ (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
+ (gst_app_src_is_seekable), (gst_app_src_check_get_range),
+ (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
+ (gst_app_src_set_caps), (gst_app_src_get_caps),
+ (gst_app_src_set_size), (gst_app_src_get_size),
+ (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
+ (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
+ (gst_app_src_set_latencies), (gst_app_src_set_latency),
+ (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
+ (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
+ * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
+ Move private data into a private instance struct. Add padding to
+ instance and class structures exposed in public headers. Add
+ Since markers to the gtk-doc blurbs (#566750).
+
+2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/app/appsrc_ex.c: Some comments.
+ Original commit message from CVS:
+ * tests/examples/app/appsrc_ex.c: (main):
+ Some comments.
+ When pulling a buffer we can get NULL when the element is EOS, don't try
+ to unref this NULL buffer.
+
+2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
+ Original commit message from CVS:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video.h:
+ Fix up build flags and include statement for the new generated
+ enumtypes files, to fix dist.
+
+2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-app.xml:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c:
+ * tests/examples/Makefile.am:
+ * tests/examples/app/Makefile.am:
+ Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
+
+2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_change_state):
+ Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
+ take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
+ this because the async_play method is deprecated and usually not called
+ anymore.
+
+2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
+ Disconnect signal handlers before destroying a previous decodebin so
+ that we don't end up causing deadlocks. Fixes #566586.
+
+2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_init),
+ (gst_audio_test_src_check_get_range),
+ (gst_audio_test_src_set_property),
+ (gst_audio_test_src_get_property):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Add property to control pull/push based scheduling.
+
+2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
+
+ Make the seek and colorkey examples depend on gtk+-x11 as they use
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/examples/seek/Makefile.am:
+ * tests/icles/Makefile.am:
+ Make the seek and colorkey examples depend on gtk+-x11 as they use
+ GDK_WINDOW_XID.
+ Fixes the build with gtk+-quartz.
+
+2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/common/: Add new exports to win32 files.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ * win32/common/libgsttag.def:
+ * win32/common/libgstvideo.def:
+ Add new exports to win32 files.
+
+2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
+ * gst-libs/gst/tag/gsttagdemux.h:
+ Add GType for GstTagDemuxResult enum.
+
+2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
+ Original commit message from CVS:
+ * gst-libs/gst/video/Makefile.am:
+ * gst-libs/gst/video/video.h:
+ Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
+ This will help bindings to use it.
+
+2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/audio/testchannels.c:
+ * win32/MANIFEST:
+ * win32/common/audio-enumtypes.c:
+ (gst_audio_channel_position_get_type),
+ (gst_ring_buffer_state_get_type),
+ (gst_ring_buffer_seg_state_get_type),
+ (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
+ * win32/common/audio-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ * win32/common/multichannel-enumtypes.h:
+ * win32/vs6/grammar.dsp:
+ * win32/vs6/libgstaudio.dsp:
+ * win32/vs7/libgstaudio.vcproj:
+ * win32/vs8/libgstaudio.vcproj:
+ Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
+ audio- in order to wrap all enums declarations of that library.
+ This modification should not matter since that header file is not a
+ public header (it will be included by public headers).
+ Modify win32 crap^Wfiles accordingly.
+
+2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Complete Sebastien's commit from the 13th by exporting the
+ _slave_method_get_type() methods.
+
+2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_init), (gst_app_src_set_property),
+ (gst_app_src_get_property), (gst_app_src_query),
+ (gst_app_src_set_latencies), (gst_app_src_set_latency),
+ (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
+ * gst-libs/gst/app/gstappsrc.h:
+ Add properties and methods to configure and retrieve the min and max
+ latencies.
+
+2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/: Implement URI query. Fixes bug #562949.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
+ (gst_gio_base_src_query):
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_query):
+ Implement URI query. Fixes bug #562949.
+
+2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Add some debug info.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (no_more_pads_cb):
+ Add some debug info.
+ * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
+ (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
+ (gst_play_sink_release_pad):
+ Add some more debug info.
+ Reconfigure the audio chain when we switch between raw and encoded audio
+ in gapless playback.
+
+2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_setcaps):
+ Pause the write thread before deactivating and releasing the ringbuffer
+ to avoid a deadlock when we do gapless playback with different sample
+ rates in playbin2. Fixes #564929.
+
+2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ Make GstAudioSrcSlaveMethod get_type() function non-static
+ as it's public now.
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstnetbuffer.def:
+ Add some missing functions to the list of exported symbols.
+
+2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
+
+ gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
+ Original commit message from CVS:
+ Patch by: Andrew Feren <acferen at yahoo dot com>
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
+ (gst_netaddress_get_address_bytes),
+ (gst_netaddress_set_address_bytes):
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ Make gst_netaddress_get_ip4_address fail for v6 addresses.
+ Make gst_netaddress_get_ip6_address either fail or return the v4
+ address as a transitional v6 address.
+ Add two convenience functions:
+ API: gst_netaddress_get_address_bytes()
+ API: gst_netaddress_set_address_bytes()
+ Fixes #564896.
+
+2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add appsrc and appsink documentation.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
+ Add appsrc and appsink documentation.
+
+2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
+ Original commit message from CVS:
+ * gst/adder/Makefile.am:
+ * gst/adder/gstadder.c:
+ Cleanup variable names to make the adder-loop easier to understand.
+ Also try to use liboil to spee it up, but ifdef it out as it does not
+ make any change for me (Intel pentim M (sse,sse2) please try on other
+ systems).
+
+2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add minimal docs to make the remaining tcp elements show up.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpserversrc.c:
+ Add minimal docs to make the remaining tcp elements show up.
+ Fixes #564139.
+
+2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Fix example to unref after emiting the push-buffer action.
+ Original commit message from CVS:
+ * examples/app/appsrc-ra.c: (feed_data):
+ * examples/app/appsrc-seekable.c: (feed_data):
+ * examples/app/appsrc-stream.c: (read_data):
+ * examples/app/appsrc-stream2.c: (feed_data):
+ Fix example to unref after emiting the push-buffer action.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
+ (gst_app_src_push_buffer_action):
+ Don't take the ref on the buffer in push-buffer action because it's too
+ awkward for bindings. Fixes #564482.
+
+2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ win32/common/config.h: Update to CVS version.
+ Original commit message from CVS:
+ * win32/common/config.h:
+ Update to CVS version.
+ * win32/common/config.h.in:
+ Hardcode path to plugin install helper exe, just like we hardcode
+ the paths in core. Removes another source of VCS conflicts for
+ people hacking gst-plugins-base on systems with autotools.
-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.
+2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-08-01 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
- * 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)
+2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-07-29 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
- Convert me to BaseTransform!! help..
+2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-29 Andy Wingo <wingo@pobox.com>
+ gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_slave_method_get_type),
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_slave_method_get_type),
+ (gst_base_audio_src_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
+ public API. This is needed for the C++ bindings to be able
+ to use this base classes. Fixes bug #564200, #564206.
+
+2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_handle_event):
+ Remove erroneous gst_buffer_ref().
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Don't forget to unref the buffer once you're done with it.
+
+2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/playback/: XRef to GstXOverlay.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ XRef to GstXOverlay.
+
+2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
+ Original commit message from CVS:
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
+ Free the factory array when finalizing.
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
+ Use a GstStaticPadTemplate since the src pad caps are fixed.
+
+2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
+ (gst_vorbis_enc_init):
+ Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
+ pad templates.
+
+2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add mapping for VP6 in avi/riff.
+
+2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
+ Original commit message from CVS:
+ * gst/subparse/samiparse.c: (sami_context_push_state),
+ (sami_context_pop_state), (start_sami_element), (end_sami_element):
+ Some versions of libxml seem to be very picky as to strict formatting
+ of the input and never 'close' the final </body> tag.
+ In order to fix that bad behaviour, we trigger the flushing of
+ remaining data on both </body> and </sami>.
+ Fixes #557365
+
+2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
+ Original commit message from CVS:
+ Patch by: Guillaume Emont <guillaume at fluendo dot com>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinders for MS Word files and OS X .DS_Store files to
+ prevent them to be recognized as MPEG files. Fixes bug #564098.
+
+2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Add some more debug info.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gen_audio_chain),
+ (gst_play_sink_reconfigure):
+ Add some more debug info.
+ Fix linking of just an encoded sink.
+ Handle failure to create a sink chain more gracefully than crashing.
+
+2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ Pushing 10 buffers is enough to run the test.
+
+2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Hook up the SKIP seek flag.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (do_seek), (stop_cb),
+ (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
+ (main):
+ Hook up the SKIP seek flag.
+
+2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (pad_added_cb):
+ Error out with a missing-plugin error when the input-selector was not
+ found.
+ * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
+ Indentation.
+
+2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
+ (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_send_event), (gst_play_sink_change_state):
+ Use G_DEFINE_TYPE.
+ Try to set the selected sink to READY before using it. This will allow
+ for detection of incompatible formats sooner.
+ Don't cause a fatal error when conversion elements are missing but post
+ a missing-element message and a warning instead because things might
+ still link and run fine.
+ Simplyfy the construction of audio and video sink chains.
+
+2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
+ (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
+ Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
+ init from glib.
+
+2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
+
+ gst/: Include glib.h instead of a specific GLib header. Including single
+ Original commit message from CVS:
+ Patch by: Luis Menina <liberforce at freeside dot fr>
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst/typefind/gsttypefindfunctions.c:
+ Include glib.h instead of a specific GLib header. Including single
+ GLib headers is deprecated. Fixes bug #563904.
+
+2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
+
+ gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
+ Original commit message from CVS:
+ 2008-12-09 Julien Moutte <julien@fluendo.com>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
+
+2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c:
+ Fix handling of odd chunks in riff metadata.
- * ext/alsa/gstalsaplugin.c (plugin_init): We are primary audio
- sinks.
+2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsasink.c (alsasink_sink_factory): Advertise our
- support of both endiannesses.
+ 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.
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Fix confusing debug message (s/event/query/)
+ 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.
-2005-07-28 Tim-Philipp Müller <tim at centricular dot net>
+2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/videotestsrc/videotestsrc.h:
- Use "_stdint.h" instead of <stdint.h>
+ 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.
-2005-07-27 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>
- * ext/vorbis/Makefile.am:
- Revert wrong commit.
+ * ChangeLog:
+ * gst/speexresample/Makefile.am:
+ fix build
+ Original commit message from CVS:
+ fix build
+
+2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Update documentation of speexresample for the new element name.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * gst/speexresample/gstspeexresample.c:
+ Update documentation of speexresample for the new element name.
+
+2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ Update README with the latest diff between the Speex resampler
+ and our copy.
+
+2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c: (plugin_init):
+ Update the debug category from speex_resample to audioresample.
+
+2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Remove audioresample files.
+ Original commit message from CVS:
+ * gst/audioresample/Makefile.am:
+ * gst/audioresample/buffer.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/debug.c:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/functable.c:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample.h:
+ * gst/audioresample/resample_chunk.c:
+ * gst/audioresample/resample_functable.c:
+ * gst/audioresample/resample_ref.c:
+ * tests/check/elements/audioresample.c:
+ Remove audioresample files.
+
+2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
+ Original commit message from CVS:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ Regenerated for library filename change.
+
+2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst/speexresample/gstspeexresample.c: (plugin_init):
+ * gst/speexresample/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/elements/speexresample.c: (setup_speexresample),
+ (GST_START_TEST), (test_pipeline):
+ Rename the moved speexresample to audioresample, integrate into the
+ build system and remove the old audioresample from the build system.
+ Fixes bug #558124, #385061, #346218, #116051.
+
+2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
+ Avoid nasty int overflows after about 12 hours and 25 minutes when these
+ code paths are triggered.
+ A free beer to Håvard Graff for finding this!
+
+2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
+ Original commit message from CVS:
+ Patch by: 이문형 <iwings at gmail dot com>
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ (gst_rtsp_connection_connect):
+ A successful gst_poll_wait() doesn't always mean successful connect() on
+ Windows. We should check errors by calling gst_poll_fd_has_error().
+ See #561924.
+
+2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
+ Original commit message from CVS:
+ * tests/check/elements/speexresample.c: (test_pipeline):
+ Make unit test again faster to prevent timeouts with valgrind.
+
+2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ Fix typo in the docs.
+
+2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
+ If no stream was found before receiving EOS, post an error message.
+ Fixes #561924.
+
+2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/: Parse segment events.
+ Original commit message from CVS:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: (gst_theora_enc_init),
+ (theora_buffer_from_packet), (theora_push_packet),
+ (theora_enc_sink_event), (theora_enc_is_discontinuous),
+ (theora_enc_chain):
+ Parse segment events.
+ Pass incomming buffer timestamps to outgoing buffers.
+ Use the running_time to construct the granulepos.
+ Fixes #562163.
+
+2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Fix buffer-duration property.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (activate_group):
+ Fix buffer-duration property.
+
+2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_change_state):
+ Really fix audiosink drain handling by keeping track of the running_time
+ of the last sample.
-2005-07-27 Wim Taymans <wim@fluendo.com>
+2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
- More compilation fixen.
+ 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.
-2005-07-27 Wim Taymans <wim@fluendo.com>
+2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_event), (gst_base_audio_sink_render),
- (gst_base_audio_sink_create_ringbuffer),
- (gst_base_audio_sink_change_state):
- Fix compilation.
-
-2005-07-27 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (setup_dynamic_link),
- (make_dv_pipeline), (make_vorbis_theora_pipeline), (query_rates),
- (query_positions_elems), (query_positions_pads), (do_seek):
- Update seek example.
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
- (gst_ogg_pad_typefind), (gst_ogg_demux_chain_elem_pad),
- (gst_ogg_demux_queue_data), (gst_ogg_demux_chain_peer),
- (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
- (gst_ogg_demux_handle_event),
- (gst_ogg_demux_deactivate_current_chain),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
- (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
- (gst_ogg_demux_chain), (gst_ogg_demux_send_event),
- (gst_ogg_demux_loop):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
- * ext/theora/theoradec.c: (theora_dec_src_event),
- (theora_dec_src_getcaps), (theora_dec_sink_event),
- (theora_dec_push), (theora_dec_chain):
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_dec_push),
- (vorbis_handle_data_packet):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_event),
- (gst_vorbisenc_chain):
- * gst/playback/gststreaminfo.c: (cb_probe):
- * gst/subparse/gstsubparse.c: (gst_subparse_src_event):
- * gst/videorate/gstvideorate.c: (gst_videorate_event):
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_event):
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame),
- (gst_ximagesink_navigation_send_event):
- * sys/xvimage/xvimagesink.c:
- (gst_xvimagesink_navigation_send_event):
- Various event updates and cleanups
+ gst-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.
-2005-07-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_prepare_images):
- Fix segfault for I420/YV12.
+ 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.
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
- Report bitrate.
+ 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.
-2005-07-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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'.
- * 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.
+2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-21 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/playback/gstplaybasebin.c: (gen_preroll_element):
- * gst/playback/gstplaybin.c: (setup_sinks),
- (gst_play_bin_change_state):
- Refcount fix and more comments.
+2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
-2005-07-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Fix random fat-fingering making this not compile.
- * sys/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.
+2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
- * 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/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.
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-11-20 22:06:05 +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.
+ 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.
-2005-07-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.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-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.
-2005-07-20 Wim Taymans <wim@fluendo.com>
+2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2005-07-19 Andy Wingo <wingo@pobox.com>
+2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
- * gst/videotestsrc/: Cleanups.
+ 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).
-2005-07-19 Wim Taymans <wim@fluendo.com>
+2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Better debugging.
+ 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.
-2005-07-19 Wim Taymans <wim@fluendo.com>
+2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ 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.
-2005-07-18 Andy Wingo <wingo@pobox.com>
+2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ Fix random type causing a docs warning.
-2005-07-18 Wim Taymans <wim@fluendo.com>
+2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
+ Original commit message from CVS:
+ * sys/v4l/gstv4l.c:
+ Give it a minimal rank for autovideosrc.
-2005-07-18 Wim Taymans <wim@fluendo.com>
+2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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.
+ 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.
- * 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.
+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().
-2005-07-16 Wim Taymans <wim@fluendo.com>
+2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/app/gstappsink.c: Fix the docs.
+ Original commit message from CVS:
+ * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
+ Fix the docs.
+
+2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
+ (gst_speex_resample_get_unit_size),
+ (gst_speex_resample_push_drain), (gst_speex_resample_event),
+ (gst_speex_resample_check_discont), (gst_speex_resample_process),
+ (gst_speex_resample_transform):
+ * gst/speexresample/gstspeexresample.h:
+ Rewrite timestamp tracking to make it more robust and guarantee
+ a continous stream.
+ * tests/check/Makefile.am:
+ * tests/check/elements/speexresample.c: (setup_speexresample),
+ (cleanup_speexresample), (fail_unless_perfect_stream),
+ (test_perfect_stream_instance), (GST_START_TEST),
+ (test_discont_stream_instance), (live_switch_alloc_only_48000),
+ (live_switch_get_sink_caps), (live_switch_push),
+ (speexresample_suite):
+ Add unit tests for speexresample based on the audioresample unit tests.
+
+2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_get_unit_size),
+ (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
+ (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
+ (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
+ (gst_speex_resample_push_drain), (gst_speex_resample_event),
+ (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
+ (gst_speex_resample_process), (gst_speex_resample_transform),
+ (gst_speex_resample_query), (gst_speex_resample_set_property):
+ * gst/speexresample/gstspeexresample.h:
+ Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
+ instead of GST_DEBUG, ...
+
+2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
+ Original commit message from CVS:
+ * gst/speexresample/gstspeexresample.c:
+ (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
+ (gst_speex_resample_process):
+ Fixate to the nearest supported rate instead of the first one.
+
+2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init), (audioresample_fixate_caps):
+ Fixate the rate to the nearest supported rate instead of
+ the first one. Fixes bug #549510.
+
+2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
+ Original commit message from CVS:
+ * gst/speexresample/README:
+ * gst/speexresample/arch.h:
+ * gst/speexresample/fixed_arm4.h:
+ * gst/speexresample/fixed_arm5e.h:
+ * gst/speexresample/fixed_bfin.h:
+ * gst/speexresample/fixed_debug.h:
+ * gst/speexresample/fixed_generic.h:
+ * gst/speexresample/resample.c: (compute_func), (main), (sinc),
+ (cubic_coef), (resampler_basic_direct_single),
+ (resampler_basic_direct_double),
+ (resampler_basic_interpolate_single),
+ (resampler_basic_interpolate_double), (update_filter),
+ (speex_resampler_init_frac), (speex_resampler_process_native),
+ (speex_resampler_magic), (speex_resampler_process_float),
+ (speex_resampler_process_int),
+ (speex_resampler_process_interleaved_float),
+ (speex_resampler_process_interleaved_int),
+ (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
+ (speex_resampler_reset_mem):
+ * gst/speexresample/speex_resampler.h:
+ Update Speex resampler with latest version from Speex GIT.
+
+2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ win32/common/libgstaudio.def: Add new symbols.
+ Original commit message from CVS:
+ * win32/common/libgstaudio.def:
+ Add new symbols.
+
+2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
+ Attempt to make obfuscated code clearer.
+
+2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Move float endianness conversion macros to core. Second part of bug ##555196.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Move float endianness conversion macros to core. Second part of
+ bug ##555196.
+
+2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/: Don't mark as gtk-doc docs as they aren't public.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.h:
+ * sys/xvimage/xvimagesink.h:
+ Don't mark as gtk-doc docs as they aren't public.
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_base_audio_sink_render):
- Align samples even if we have roundoff errors in the
- timestamp conversion.
+2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-07-16 Wim Taymans <wim@fluendo.com>
+ 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
- * 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.
+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>
- * 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.
+ 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.
- * gst-libs/gst/audio/gstbaseaudiosrc.c:
- (gst_base_audio_src_get_time):
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire),
- (gst_ring_buffer_set_sample), (gst_ring_buffer_commit),
- (gst_ring_buffer_read):
- * gst-libs/gst/audio/gstringbuffer.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_get_times),
- (gst_ximagesink_show_frame):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
- Correcly convert buffer timestamp to stream time.
+2008-10-16 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.
-2005-07-16 Wim Taymans <wim@fluendo.com>
+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/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_get_buffer):
- Timestamp buffers correctly.
+2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/playback/gstplaybin.c: (gen_video_element):
- Make internal fakesink silent.
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Fix debug statements (space between '%' and actual format).
-2005-07-15 Wim Taymans <wim@fluendo.com>
+2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/ffmpegcolorspace/Makefile.am:
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info),
- (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
- (gst_ffmpegcsp_get_type), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_get_size),
- (gst_ffmpegcsp_transform_ip), (gst_ffmpegcsp_transform):
- Ported ffmpegcolorspace to basetransform.
+ gst/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.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_transform):
- * gst/volume/gstvolume.c: (volume_transform):
- Ported to new API.
+2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
-2005-07-14 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ changelog
+ Original commit message from CVS:
+ changelog
+
+2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
+ Original commit message from CVS:
+ 2008-10-08 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
+ target instead of setting it.
+ (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
+ API for a decode pad. The bugfix is that we set the group in
+ activate(), not when the pad was created because it might be NULL
+ then.
+ (gst_decode_group_control_source_pad, gst_decode_group_expose):
+ Update to use the API.
+
+2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
+ Original commit message from CVS:
+ 2008-10-08 Andy Wingo <wingo@pobox.com>
+ * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
+ be a subclass of GstGhostPad.
+ (analyze_new_pad): So, when emitting the signals that determine
+ how we do autoplugging, already create the ghost pad and use it as
+ the pad in the signal arguments. This allows applications to make
+ a connection between the pad passed in e.g. autoplug-continue, and
+ the pad passed in new-decoded-pad.
+ (connect_pad, expose_pad): Update to receive the ghosted decode
+ pad in the args, retargetting it as necessary if we have to plug
+ the target pad through a multiqueue.
+ (gst_decode_group_control_source_pad): Adapt to receive an
+ already-ghosted pad that just needs activation, blocking, and
+ drain notification.
+ (sort_end_pads): Adapt for decode pads actually being pads.
+ (gst_decode_group_expose): Adapt for decode pads actually being
+ pads. Rewrite the decode pad names so they appear in order. Adds a
+ new error case if we couldn't set the name.
+ (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
+ logic.
+ (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
+ New API for the decode pad, needed because we shouldn't do these
+ things inside gst_decode_pad_new(), but after.
+ (gst_decode_pad_new): Change to actually make the real pad, and
+ delay the blocking/drainage bits.
+
+2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
+
+ ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
+ Original commit message from CVS:
+ Patch by: Daniel Drake <dsd at laptop dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
+ Unref all buffers when clearing collectpads. Fixes bug #546955.
+
+2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
+
+ ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
+ Original commit message from CVS:
+ Based on a patch by: Klaas <klaas at rivercrew dot net>
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
+ (gst_vorbis_enc_buffer_check_discontinuous),
+ (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
+ * ext/vorbis/vorbisenc.h:
+ Keep track of the upstream segments and use the running time on that
+ segment instead of the buffer timestamp everywhere. Fixes bug #525807.
+
+2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (audio_convert_convert):
+ Prevent overflows with big buffer when calculating the size of
+ the intermediate buffer by using gst_util_uint64_scale() instead of
+ plain arithmetics. Fixes bug #552801.
+
+2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
+
+ ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
+ Original commit message from CVS:
+ Patch by: Pavel Zeldin <pzeldin at gmail dot com>
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
+ (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
+ (gst_clock_overlay_init), (gst_clock_overlay_set_property),
+ (gst_clock_overlay_get_property):
+ * ext/pango/gstclockoverlay.h:
+ API: Add ability to specify format for date/time display by
+ adding a "time-format" property.
+ Fixes bug #554879.
+
+2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
+
+ gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
+ Original commit message from CVS:
+ Patch by: Jan Gerber <j at oil21 dot org>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add FFV1 fourcc to support playback of FFMPEG lossless video
+ in AVI. Fixes bug #555319.
+
+2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
+ Original commit message from CVS:
+ Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Implement skew clock slaving. Fixes #552559.
+
+2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/: Fix include of config.h
+ Original commit message from CVS:
+ * gst-libs/gst/audio/multichannel.c:
+ * gst-libs/gst/audio/testchannels.c:
+ Fix include of config.h
+
+2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
+ Original commit message from CVS:
+ Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
+ (print_media), (gst_sdp_message_dump):
+ Fix parsing of the c= field containing multicast addresses.
+ Fixes #552199.
+ Add the connection info to the session or streams.
+ Fix parsing of the bandwidth.
+ Add debugging for the connections and bandwidths for a media.
+ Add debugging for the bandwidth of the session.
+
+2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_change_state):
+ Configure the next seqnum and timestamp in the state change so that they
+ can be queried soon after.
+
+2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain):
+ Improve debugging of the rtptime.
+
+2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to development -> 0.10.21.1
+ Original commit message from CVS:
+ * configure.ac:
+ Back to development -> 0.10.21.1
+
+2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/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.
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
- Also draw image when not from a pool.
+2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-14 Wim Taymans <wim@fluendo.com>
+ 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/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
- (mute_stream), (silence_stream):
- Small debug additions.
+2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-07-14 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose),
- (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page),
- (gst_ogg_demux_activate_chain), (gst_ogg_demux_loop):
- Better error recovery, ignore unconnected pads and
- non-fatal errors.
+2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-07-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/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
+=== release 0.10.21 ===
-2005-07-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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
+ * 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
+
+2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-07-13 Andy Wingo <wingo@pobox.com>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/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
- * sys/v4l/gstv4lsrc.c (gst_v4lsrc_stop): Fix a spurious warning.
- (gst_v4lsrc_fixate): Fixate on format as well.
+2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ configure.ac: 0.10.20.4 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.20.4 pre-release
-2005-07-12 Andy Wingo <wingo@pobox.com>
+2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
- * 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.
+ ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
+ Original commit message from CVS:
+ Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
+ * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
+ Set the BOS flag on the BOS packet. Fixes #553244.
-2005-07-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ (gst_rtsp_message_parse_request),
+ (gst_rtsp_message_parse_response):
+ Fix the g_return_val_if_fail() statements.
-2005-07-10 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
- * 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>
+ 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
- * 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.
+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).
- * sys/v4l/v4lsrc_calls.h:
- * sys/v4l/v4lsrc_calls.c: Remove sync-related stuff.
- (gst_v4lsrc_get_fps_list): Moved here from gstv4lsrc.c.
- (gst_v4lsrc_buffer_new): Totally derive from GstBuffer.
+2008-09-13 11:04:02 +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.
+ 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).
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/gstv4lxoverlay.c:
- * sys/v4l/gstv4ltuner.h:
- * sys/v4l/gstv4ltuner.c: Header loc fixen.
-
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/gstv4lsrc.c: Crucial GPL update. Clean up a bit, port to
- PushSrc/BaseSrc. Removed most sync-related properties, videorate
- or something should handle that. Made a live source.
+2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lelement.c: Derive from GstPushSrc. No more
- signals. Some cleanups.
+ 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.
- * sys/v4l/gstv4lcolorbalance.h: Interface header update.
+2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/v4l/gstv4l.c: Don't register v4lelement, or the jpeg/mjpeg
- stuff.
+ 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.
- * sys/v4l/Makefile.am: Build everything except the jpeg/mjpeg
- stuff.
+2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/Makefile.am (SUBDIRS): Hit the V4L crack pipe.
+ 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).
-2005-07-07 Wim Taymans <wim@fluendo.com>
+2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2005-07-06 Wim Taymans <wim@fluendo.com>
+2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2005-07-06 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.
- * ext/theora/theoradec.c: (theora_dec_src_getcaps),
- (theora_dec_push), (theora_handle_data_packet):
- Prepare for better timestamp fix later.
+2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioconvert/gstaudioconvert.c:
- List most accurate caps first
+ 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.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
- Use proper pad task function.
+2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_show_frame):
- Fix deadlock when alloc failed.
+ 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.
-2005-07-05 Andy Wingo <wingo@pobox.com>
+2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/gnomevfs/gstgnomevfssrc.c:
- * gst/sine/gstsinesrc.c:
- * gst/tcp/gsttcpserversrc.c:
- * gst/tcp/gsttcpclientsrc.c: s/BASESRC/BASE_SRC/.
+ 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
- * sys/v4l/: Port from 0.8.
+2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
- * Many files: Null if we got it....
+ 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.
-2005-07-05 Andy Wingo <wingo@pobox.com>
+2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
- * gst/tcp/gsttcpserversink.c (gst_tcpserversink_handle_server_read):
- * gst/tcp/gstmultifdsink.c (gst_multifdsink_client_queue_data):
- Signedness fixes.
+ 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.
-2005-07-05 Wim Taymans <wim@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.
- * 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>
+2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
- * gst/videoscale/gstvideoscale.c:
- * gst/videoscale/gstvideoscale.h: Clean up, port to 0.9. Derives
- from BaseTransform, implements a transform_caps. Removed dead code
- including some PAR stuff that was never reached -- should probably
- be added back somehow.
+ gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
+ check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
+ Fixes #548065.
-2005-07-01 Andy Wingo <wingo@pobox.com>
+2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/videoscale: Merge David's work from 0.8 branch. Changes to
- come later.
+ 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".
-2005-06-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.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
+ 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.
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- Fix build after riff changes.
+ 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.
-2005-06-30 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps),
- (gst_riff_create_iavs_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst-libs/gst/riff/riff-read.h:
- * gst-libs/gst/riff/riff.c: (gst_riff_init):
- Add gst_riff_init() to initialize the debug category, instead
- of plugin_init(). Port riff-media.[ch] from -THREADED to HEAD.
+ gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ When not slaved to another clock also subtract the base_time from our
+ internal clock time to get the running time.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init):
- Oops, I shouldn't apply hacks.
+ 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.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init):
- Remove pad_loop function which doesn't work.
+ 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.
-2005-06-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.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.
+ 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.
-2005-06-29 Wim Taymans <wim@fluendo.com>
+2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
- (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ringbuffer_set_callback):
- Fix compilation error.
- Ringbuffer starts out as not running.
- Free our clock in dispose.
- When releasing the ringbuffer we need to renegotiate so
- clear the pad caps.
+ gst-libs/gst/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.
-2005-06-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * autogen.sh:
- * configure.ac:
- * docs/Makefile.am:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-libs-docs.sgml:
- * docs/libs/gst-plugins-libs-sections.txt:
- * docs/libs/gst-plugins-libs.types:
- * docs/libs/tmpl/gstaudio.sgml:
- * docs/libs/tmpl/gstcolorbalance.sgml:
- * docs/libs/tmpl/gstringbuffer.sgml:
- * gst-libs/gst/audio/gstringbuffer.c:
- (gst_ringbuffer_set_callback):
- reinstate gtk-doc docs for plugin libs
+ 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.
-2005-06-28 Wim Taymans <wim@fluendo.com>
+2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
- (gst_ogg_demux_init):
- Removed pad loop function.
+ gst-libs/gst/: Reducing number of dundocumented symbols.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ * gst-libs/gst/tag/gsttagdemux.h:
+ Reducing number of dundocumented symbols.
-2005-06-28 Wim Taymans <wim@fluendo.com>
+2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ 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.
-2005-06-28 Andy Wingo <wingo@pobox.com>
+2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * *.c: Don't cast to GstObject before reffing/unreffing.
+ ext/vorbis/vorbisdec.c: Do not leak old taglist.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c:
+ Do not leak old taglist.
-2005-06-27 Andy Wingo <wingo@pobox.com>
+2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/videotestsrc/gstvideotestsrc.c
- (gst_videotestsrc_activate_push): Activation API changes.
+ 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().
- * 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.
+2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
- * 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.
+ gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
+ Original commit message from CVS:
+ 2008-08-04 Andy Wingo <wingo@pobox.com>
+ * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
+ documentation fix.
-2005-06-27 Edward Hervey <edward@fluendo.com>
+2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/theora/theoradec.c: (theora_dec_change_state):
- re-arranged call to parent's state change in order to avoid locks (or
- worse).
+ gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Cleanup lots of empty lines that came from gst-indent going havoc
+ before I added the INDENT_ON/OFF marker some time agao.
-2005-06-26 Edward Hervey <edward@fluendo.com>
+2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
- 2nd argument of 'unknow-type' signal is a GstCaps and not a
- GstMiniObject
+ Bump requirement to latest core and use new tag for riff formats.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/riff/riff-read.c:
+ Bump requirement to latest core and use new tag for riff formats.
+ Needed for #520694.
-2005-06-25 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
- Set the worker thread's running flag to TRUE before starting the
- thread.
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init):
- Catch a failure to add typefind to the bin.
+2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-06-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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'.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init), (gst_sinesrc_create),
- (gst_sinesrc_set_property), (gst_sinesrc_get_property),
- (gst_sinesrc_start):
- * gst/sine/gstsinesrc.h:
- add num-buffers and timestamp-offset properties
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_set_property),
- (gst_videotestsrc_get_property):
- add timestamp-offset property
+2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-06-23 Christian Schaller <uraeus@gnome.org>
+ 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.
- * configure.ac: add videorate
- * gst-plugins-base.spec.in: add videorate
+2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-06-23 Wim Taymans <wim@fluendo.com>
+ 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/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_event), (gst_videorate_chain):
- Fixed videorate, fixating an already fixated caps is not
- an error.
+2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-06-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/ogg/README:
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_set_header_on_caps):
- Buffer on caps is not boxed anymore.
+2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-06-22 Wim Taymans <wim@fluendo.com>
+ gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Dist recently-added gstfastrandom.h.
- * 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.
+2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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 (?).
- * configure.ac:
- back to HEAD
+2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-=== release 0.9.1 ===
+ gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Adding acid chunk for tempo and loop information.
-2005-06-19 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * .cvsignore:
- * NEWS:
- * README:
- * RELEASE:
- * configure.ac:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
- updates for release
+ sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
+ Original commit message from CVS:
+ * sys/xvimage/Makefile.am:
+ floor() needs linking to $(LIBM).
-2005-06-09 Andy Wingo <wingo@pobox.com>
+2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
-
-2005-06-09 Andy Wingo <wingo@pobox.com>
+ 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
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/net/Makefile.am:
- Add gstnet to build.
+2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-06-09 Andy Wingo <wingo@pobox.com>
+ 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-libs/gst/gconf/gconf.c:
- * gst/playback/test.c:
- * gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
- fixes.
+2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioconvert/gstaudioconvert.c: RPAD fixes.
+ 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.
- * ext/theora/theoraenc.c (theora_enc_chain):
- * ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.
+2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
- * ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
- RealPad.
+ 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.
-2005-06-02 Wim Taymans <wim@fluendo.com>
+2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst-libs/gst/net/Makefile.am:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs.pc.in:
- Added net stuff, version net lib.
+ ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
+ Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
+ This fixes a critical warning.
-2005-06-02 Wim Taymans <wim@fluendo.com>
+2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * examples/seeking/seek.c: (make_vorbis_theora_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (do_seek):
- Updated seek example.
+ ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Allow muxing of CELT into Ogg streams.
-2005-06-02 Andy Wingo <wingo@pobox.com>
+2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * pkgconfig/gstreamer-libs-uninstalled.pc.in (prefix):
- * pkgconfig/gstreamer-libs.pc.in (prefix): Add gst/tag to the -L
- list.
+ 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).
- * 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.
+2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
-2005-06-02 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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>
+2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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>
+ 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/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.
+2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioquantize.c:
+ (gst_audio_quantize_setup_dither),
+ (gst_audio_quantize_free_dither):
+ * gst/audioconvert/gstfastrandom.h:
+ Implement a linear congruential generator as pseudo random number
+ generator for the dither noise. This is about 2 times faster than
+ using GLib's mersenne twister. Also this uses only integer math for
+ generating integers while GLib internally uses floating point math.
+
+2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
+
+ configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
+ Original commit message from CVS:
+ * configure.ac:
+ Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
-2005-05-25 Wim Taymans <wim@fluendo.com>
+2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * 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-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien.lespiau gmail com>
+ * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
+ Use GST_STR_NULL to avoid crashes with libcs that don't
+ like NULL strings in printf args (such as the win32 one).
+ Fixes #544306.
-2005-05-25 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/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_create):
- Fix memleaks, GST_BUFFER_DATA() is not freed.
+2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
-2005-05-25 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.
- * ext/alsa/gstalsasink.c: (gst_alsasink_open):
- Open non-blocking, set to blocking mode afterwards to avoid
- lockups when audio device is busy.
+2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make it impossible to have NULL caps at the point where we set
+ framerate and other things. Also don't return immediately for "3ivd"
+ video and let framerate, etc be set. Might fix bug #542508.
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
- This can't be good.
+2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-2005-05-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
+ Original commit message from CVS:
+ * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
+ Video format can also be conveniently determined from (many)
+ non-fixed caps.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_chain), (gst_audio_convert_link_src),
- (gst_audio_convert_setcaps):
- Implement instant setup switching.
+2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Fix missing unlock.
- * gst/playback/gstplaybin.c: (add_sink):
- First add, then link (otherwise pad link fails).
+2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * examples/Makefile.am:
- fix buildbot (make distcheck)
+2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/alsa/gstalsamixer.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisparse.c:
+ * ext/vorbis/vorbistag.c:
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstqueue2.c:
+ * gst/playback/gsturidecodebin.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/volume/gstvolume.c:
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ Cleanup Plugin docs. Link to signals and properties. Fix sub-section
+ titles. Drop mentining that all our example pipelines are "simple"
+ pipelines.
-2005-05-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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/playback/gstplaybin.c: (gen_vis_element):
- Remove some wrong code. Doesn't work yet.
+2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
+ Original commit message from CVS:
+ * tests/examples/seek/Makefile.am:
+ Fix out of tree build by adding all required CFLAGS.
+
+2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_raw_queue):
+ And ref the pad before returning it again when linking to the queue
+ failed. Otherwise we will unref the pad twice later and things break.
+
+2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_raw_queue):
+ If linking the raw pad with a queue fails, try it without a queue
+ instead of failing completely. This should never happen.
+
+2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
+
+ gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
+ Original commit message from CVS:
+ Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
+ * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
+ Add a queue after a demuxer if the demuxer outputs raw data. This was
+ done before only for non-raw data but is required in this case too.
+ Fixes bug #540215.
+ decodebin2 doesn't have this issue because all streams of a group
+ go through multiqueue.
+
+2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
+
+ gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Makes libgstsdp compile with mingw32 by defining the right WINVER so
+ that getaddrinfo() can be used. Fixes #541358.
+
+2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_init),
+ (gst_video_test_src_set_property),
+ (gst_video_test_src_get_property), (gst_video_test_src_create):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Cleanups, use default property values as defines.
+ Add property to enable/disable peer buffer allocation.
+
+2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
+ * tests/check/pipelines/streamheader.c: (streamheader_suite):
+ Enable unit tests on PPC again as the bugs are now fixed.
+
+2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
+ Fixes bug #540351.
+
+2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_get_unit_size):
+ Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
+ it on other formats. Also adjust the unit size only for that format
+ to not include the palette. Fixes bug #540497.
+
+2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
-2005-05-19 Wim Taymans <wim@fluendo.com>
+2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/net/Makefile.am:
- * gst-libs/gst/net/README:
- * gst-libs/gst/net/gstnetbuffer.c: (gst_netbuffer_get_type),
- (gst_netbuffer_class_init), (gst_netbuffer_init),
- (gst_netbuffer_finalize), (gst_netbuffer_copy),
- (gst_netbuffer_new), (gst_netaddress_set_ip4_address),
- (gst_netaddress_set_ip6_address), (gst_netaddress_get_net_type),
- (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address):
- * gst-libs/gst/net/gstnetbuffer.h:
- Added buffer subclass to store extra to/from addresses for
- network sources/sinks.
+ ChangeLog: ChangeLog surgery.
+ Original commit message from CVS:
+ * ChangeLog:
+ ChangeLog surgery.
+ * tests/examples/seek/seek.c:
+ Move variable into ifdef too.
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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().
- * gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
- Don't lock an unassigned variable.
+2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Name the test registry format neutral.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Name the test registry format neutral.
- * 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.
+2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-18 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-17 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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:
+2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ChangeLog: Forgot to save the ChangeLog :/
+ Original commit message from CVS:
+ * ChangeLog:
+ Forgot to save the ChangeLog :/
- * 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.
+2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/: Embedd the xwindow.
+ Original commit message from CVS:
+ * tests/examples/seek/Makefile.am:
+ * tests/examples/seek/seek.c:
+ Embedd the xwindow.
+
+2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
+ (gst_ximagesink_setcaps):
+ * sys/ximage/ximagesink.h:
+ When the caps change, make sure to re-draw borders in
+ force-aspect-ratio=true mode.
+ * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
+ Don't clear the border_draw flag until we actually draw the border.
+ * tests/check/Makefile.am:
+ Ignore alsasink/src during the states test too, so it doesn't fail
+ when running without access to the sound device.
+
+2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Fix crasher when playing a parse-launch line the 2nd time.
-2005-05-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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>
+ 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.
- * gst/volume/Makefile.am:
- * gst/volume/demo.c
- * gst/volume/gstvolume.h
- * gst/volume/gstvolume.c: Port to 0.9 API, derive from
- basetransform. Probably need an audio filter base class.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
- (gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
- (gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
- (gst_vorbisenc_chain):
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- Make caps writable before writing to it.
- Fix negotiation in audioconvert some more.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_event), (gst_videorate_chain):
- Better negotiation.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_getcaps), (gst_videorate_setcaps),
- (gst_videorate_blank_data), (gst_videorate_init),
- (gst_videorate_event), (gst_videorate_chain),
- (gst_videorate_change_state):
- Port videorate, do a better job at negotiation while we're at
- it.
-
-2005-05-09 Jan Schmidt <thaytan@mad.scientist.com>
+2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- Disable libvisual
+ * ChangeLog:
+ break long lines
+ Original commit message from CVS:
+ break long lines
- * examples/Makefile.am:
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/riff/Makefile.am:
- * gst-libs/gst/tag/Makefile.am:
- * gst-libs/gst/video/Makefile.am:
- Fixups for missing variables.
-
-2005-05-09 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_theora_pipeline),
- (make_vorbis_theora_pipeline), (make_avi_msmpeg4v3_mp3_pipeline),
- (query_rates), (query_positions_elems), (query_positions_pads),
- (update_scale), (play_cb), (pause_cb), (stop_cb), (main):
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
- (gst_ogg_pad_query_types), (gst_ogg_pad_src_query),
- (gst_ogg_pad_typefind), (gst_ogg_demux_init),
- (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
- (gst_ogg_demux_read_end_chain), (gst_ogg_demux_sink_activate):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_push_page), (gst_ogg_mux_queue_pads),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_collected), (gst_ogg_mux_change_state):
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
- (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_sink_query), (gst_ogm_parse_chain):
- * ext/theora/theoradec.c: (gst_theora_dec_init), (_inc_granulepos),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_sink_query),
- (theora_dec_src_event), (theora_dec_sink_event),
- (theora_handle_comment_packet), (theora_handle_type_packet),
- (theora_handle_header_packet), (theora_handle_data_packet),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_sink_query), (vorbis_dec_src_event),
- (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
- (vorbis_handle_type_packet), (vorbis_handle_header_packet),
- (copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
- (gst_vorbisenc_sink_query), (gst_vorbisenc_init),
- (gst_vorbisenc_sink_event), (gst_vorbisenc_chain):
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (gst_play_bin_query):
- * gst/playback/test3.c: (update_scale):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_setcaps),
- (gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start):
- * gst/subparse/gstsubparse.c: (gst_subparse_init):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
- (gst_videotestsrc_src_query):
- * gst/videotestsrc/videotestsrc.c: (paint_hline_I420),
- (paint_hline_Y41B), (paint_hline_Y42B), (paint_hline_Y800),
- (paint_hline_YUV9):
- * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame):
- Port to new query API.
- Updated seek.
- Cleanups in x[v]imagesink
-
-2005-05-09 Andy Wingo <wingo@pobox.com>
+2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
- * ext/alsa/gstalsasink.h:
- * ext/gnomevfs/gstgnomevfssrc.c:
- (gst_gnomevfssrc_get_icy_metadata):
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_perform_seek)
- (gst_ogg_demux_read_chain, gst_ogg_demux_read_end_chain)
- * ext/theora/theoradec.c (theora_dec_src_query)
- (theora_dec_src_event, theora_dec_sink_event)
- (theora_handle_comment_packet, theora_handle_data_packet):
- * ext/theora/theoraenc.c (theora_enc_chain):
- * ext/vorbis/vorbisdec.c (vorbis_dec_src_event)
- (vorbis_dec_sink_event, vorbis_handle_comment_packet):
- * gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
- * gst/typefind/gsttypefindfunctions.c (mp3_type_find)
- (qt_type_find):
- * gst/videotestsrc/videotestsrc.c (paint_setup_I420)
- (paint_setup_YV12, paint_setup_YUY2, paint_setup_UYVY)
- (paint_setup_YVYU, paint_setup_IYU2, paint_setup_Y41B)
- (paint_setup_Y42B, paint_setup_Y800, paint_setup_IMC1)
- (paint_setup_IMC2, paint_setup_IMC3, paint_setup_IMC4)
- (paint_setup_YVU9, paint_setup_YUV9, paint_setup_xRGB8888)
- (paint_setup_xBGR8888, paint_setup_RGBx8888)
- (paint_setup_BGRx8888, paint_setup_RGB888, paint_setup_BGR888)
- (paint_setup_RGB565, paint_setup_xRGB1555):
- * gst/videotestsrc/videotestsrc.h:
- * sys/ximage/ximagesink.c (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c (gst_xvimagesink_get_xv_support)
- (gst_xvimagesink_setcaps, gst_xvimagesink_buffer_alloc):
- GCC4 fixes.
-
- * ext/ogg/gstoggdemux.c (gst_ogg_demux_find_chains): Use the new
- gst_pad_query_position. Fixes oggdemux.
-
-2005-05-08 David Schleef <ds@schleef.org>
-
- * configure.ac: Require liboil.
- * gst/videotestsrc/gstvideotestsrc.c: Fix up liboil calls, add
- a few more.
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
+ 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.
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
- (gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
- Well, unreffing a buffer right before pushing it is asking
- for trouble..
+ 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.
-2005-05-06 Christian Schaller <uraeus@gnome.org>
+2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
- * pkgconfig/gstreamer-libs.pc.in: add missing library calls
+ gst/playback/: Fix a whole bunch of typos in comments and log statements.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ Fix a whole bunch of typos in comments and log statements.
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.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.
+ 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.
+2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
-2005-05-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst-libs/gst/audio/gstaudiosink.c:
- (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
- (audioringbuffer_thread_func), (gst_audioringbuffer_init),
- (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
- (gst_audioringbuffer_play), (gst_audioringbuffer_stop),
- (gst_audioringbuffer_delay), (gst_audiosink_class_init),
- (gst_audiosink_create_ringbuffer):
- * gst-libs/gst/audio/gstbaseaudiosink.c:
- (gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
- (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
- (gst_baseaudiosink_set_property), (build_linear_format),
- (debug_spec_caps), (debug_spec_buffer),
- (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
- (gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
- (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
- (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
- * gst-libs/gst/audio/gstbaseaudiosink.h:
- * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
- (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
- (gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
- (gst_ringbuffer_play), (gst_ringbuffer_pause),
- (gst_ringbuffer_stop), (gst_ringbuffer_delay),
- (gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
- (wait_segment), (gst_ringbuffer_commit),
- (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
- (gst_ringbuffer_clear):
- * gst-libs/gst/audio/gstringbuffer.h:
- Make the base audiosink return an error when there is no
- audiobuffer negotiated.
+2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2005-05-06 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ * gst/tcp/README:
+ apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
+ Original commit message from CVS:
+ apparently it's an error to specify nc -l -p 3000 - though the short usage
+ does not make it very clear that you can drop the host arg with -l
+
+2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
+ (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
+ Report the encoder latency. Fixes #538232.
+
+2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
+ (notify_source), (activate_group):
+ Implement the source property, emit notify when it changes in the
+ underlying uridecodebin.
+
+2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (stop_cb):
+ Free and clear the seek element list so that we don't use invalid
+ references when seeking after recreating a gst-launch line.
+
+2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_render):
+ Report latency even if we are not live instead of hiding it.
+ Take ts-offset and render-delay of the basesink into account when
+ scheduling samples.
+ Rework the clipping code so that we can take the various offsets into
+ account and still do correct clipping.
+
+2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Bump verion back to devel -> 0.10.20.1
+ Original commit message from CVS:
+ * configure.ac:
+ Bump verion back to devel -> 0.10.20.1
+
+2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
+ Don't increase the size of non-string image buffers by one as this
+ might in theory confuse decoders. Still increase it by one for string
+ image buffers to append '\0'.
+
+2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
+
+ gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
+ Fix a buffer memleak and remove a confusing and wrong debug output.
+ Fixes bug #538663.
+
+2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/appsink-src.c: Don't use a buffer after unreffing it.
+ Original commit message from CVS:
+ * examples/app/appsink-src.c: (on_new_buffer_from_source):
+ Don't use a buffer after unreffing it.
- * ext/Makefile.am:
- Disable cdparanoia until someone ports it!
+=== release 0.10.20 ===
-2005-05-06 Wim Taymans <wim@fluendo.com>
+2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
- (gst_ogg_demux_sink_activate):
- And revert after wingo's revert.. sigh..
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * po/LINGUAS:
+ * win32/common/config.h:
+ Release 0.10.20
+ Original commit message from CVS:
+ Release 0.10.20
-2005-05-05 Andy Wingo <wingo@pobox.com>
+2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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/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: Foo
+ * 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.
-2005-05-05 Wim Taymans <wim@fluendo.com>
+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.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_caps_remove_format_info),
- (gst_audio_convert_setcaps), (gst_audio_convert_fixate),
- (gst_audio_convert_change_state), (gst_audio_convert_channels):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
- (gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
- (gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
- (gst_videotestsrc_init), (gst_videotestsrc_loop):
- Don't ignore _push() return values.
- Make sure no processing is done when shutting down.
- Videotestsrc pad activation fix.
+2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Do not use short_description in section docs for elements. We extract them from element details and there will be war...
+ Original commit message from CVS:
+ * ext/dc1394/gstdc1394.c:
+ * ext/ivorbis/vorbisdec.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/metadata/gstmetadatademux.c:
+ * ext/mythtv/gstmythtvsrc.c:
+ * ext/theora/theoradec.c:
+ * gst-libs/gst/app/gstappsink.c:
+ * gst/bayer/gstbayer2rgb.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/rawparse/gstaudioparse.c:
+ * gst/rawparse/gstvideoparse.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpclient.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/selector/gstinputselector.c:
+ * gst/selector/gstoutputselector.c:
+ * gst/videosignal/gstvideoanalyse.c:
+ * gst/videosignal/gstvideodetect.c:
+ * gst/videosignal/gstvideomark.c:
+ * sys/oss4/oss4-mixer.c:
+ * sys/oss4/oss4-sink.c:
+ * sys/oss4/oss4-source.c:
+ Do not use short_description in section docs for elements. We extract
+ them from element details and there will be warnings if they differ.
+ Also fixing up the ChangeLog order.
+
+2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.19.3 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.19.3 pre-release
+
+2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ Fix build on win32.
+ Patch By: David Schleef <ds@schleef.org>
+ Fixes: #536874
+
+2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
+ (gst_gio_base_src_create):
+ * ext/gio/gstgiobasesrc.h:
+ Try to read the requested number of bytes, even if the first
+ read returns less than requested, until nothing is read anymore
+ or we have the requested amount of bytes. This fixes playback of
+ files via Samba as Samba only allows to read 64k at once.
+ Implement a caching algorithm that makes sure that we read at
+ least 4k of data every time. Some elements will try to read a few
+ bytes, then seek, read again a few bytes and so on and this is
+ painfully slow as every operation has to go over DBus if GVfs is
+ used as backend.
+ Fixes bug #536849 and #536848.
+ * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
+ (gst_gio_src_check_get_range):
+ Override check_get_range() to blacklist http/https URIs
+ and whitelist file URIs. More to be added on demand.
+
+2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
+ Original commit message from CVS:
+ * examples/app/Makefile.am:
+ * examples/app/appsrc-ra.c: (feed_data), (seek_data),
+ (found_source), (bus_message), (main):
+ * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
+ (found_source), (bus_message), (main):
+ * examples/app/appsrc-stream2.c: (feed_data), (found_source),
+ (bus_message), (main):
+ Added 3 more example application for using appsrc in random-access mode,
+ pull-mode streaming and pull mode seekable.
+ * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
+ (gst_app_src_start), (gst_app_src_do_get_size),
+ (gst_app_src_create):
+ * gst-libs/gst/app/gstappsrc.h:
+ Make stream-type property writable.
+ Unset flushing when starting so that we reuse appsrc.
+ Inform basesrc about the configured size.
+ Emit seek-data signal when we are going to a different offset in
+ random-access mode.
+
+2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
+ Original commit message from CVS:
+ * examples/app/appsrc-stream.c: (found_source), (main):
+ Use deep-notify until we can depend on a playbin2 with support for the
+ source property.
+
+2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
+ Original commit message from CVS:
+ * examples/app/.cvsignore:
+ * examples/app/Makefile.am:
+ * examples/app/appsrc-stream.c: (read_data), (start_feed),
+ (stop_feed), (found_source), (bus_message), (main):
+ Added an example on how to use appsrc in playbin in streaming mode from
+ an mmapped file.
+ * examples/app/appsrc_ex.c: (main):
+ Set pipeline to NULL to free queued buffers.
+ * gst-libs/gst/app/gstapp-marshal.list:
+ * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
+ (gst_app_src_class_init), (gst_app_src_init),
+ (gst_app_src_flush_queued), (gst_app_src_dispose),
+ (gst_app_src_set_property), (gst_app_src_get_property),
+ (gst_app_src_unlock), (gst_app_src_unlock_stop),
+ (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
+ (gst_app_src_check_get_range), (gst_app_src_do_seek),
+ (gst_app_src_create), (gst_app_src_set_stream_type),
+ (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
+ (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
+ (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
+ (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
+ (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
+ * gst-libs/gst/app/gstappsrc.h:
+ Measure max queue size in bytes instead.
+ Add support for 3 modes of operation, streaming, seekable and
+ random-access, making basesrc handle the scheduling modes for each.
+ Add appsrc:// uri handler so that automatic plugging can be done from
+ playbin2 or uridecodebin, for example.
+ Added support for custom segment formats.
+ Add support for push and pull based operations from the application.
+ Expand the methods so that errors can be detected.
+ Flush the queued buffers on seeks and when shutting down.
+ Add signals to inform the app that a seek must happen.
+
+2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: 0.10.19.2 pre-release
+ Original commit message from CVS:
+ * configure.ac:
+ 0.10.19.2 pre-release
+
+2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ win32/common/: Add new API functions to the dll exports
+ Original commit message from CVS:
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgsttag.def:
+ Add new API functions to the dll exports
+
+2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ Disconnect signals from decodebins we created before we remove it from
+ playbin, to avoid crashes if the decodebin is eventually disposed after
+ the playbin itself (possible if the app takes a reference on the
+ decodebin).
+ Fixes #536521.
-2005-05-05 Wim Taymans <wim@fluendo.com>
+2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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>
+ 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/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.
+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.
-2005-04-28 Wim Taymans <wim@fluendo.com>
+2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
+
+ gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
+ Don't set caps on the buffers that contain a copy of the buffer
+ including the caps of them resulting in an always increasing refcount
+ of the caps and insanely large caps. Instead include a buffer without
+ caps in the new caps. Fixes bug #536475.
+
+2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
+ Transform a given PAR to a range on the struct with the generic
+ height/width instead of the struct with the possibly restricted
+ height/width.
+
+2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
+ Prefer the given format if it contains something stricter than [1,MAX]
+ for height or width and only put a structure that requires rescaling
+ as second. This makes it possible to use videoscale in pipelines where
+ the source can actually produce the wanted height/width but usually
+ selects a different one from the requested.
+
+2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
+
+ gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
+ Original commit message from CVS:
+ Based on patch by: John Millikin <jmillikin gmail com>
+ * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
+ (gst_vorbis_tag_add_coverart):
+ Retrieve COVERART tags from vorbis comments (#512333)
- * ext/theora/theoradec.c: (_inc_granulepos),
- (theora_dec_sink_event), (theora_handle_comment_packet),
- (theora_handle_type_packet), (theora_handle_header_packet),
- (theora_handle_data_packet), (theora_dec_chain),
- (theora_dec_change_state):
- Refactor a bit, use STREAM_LOCK.
+2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-28 Wim Taymans <wim@fluendo.com>
+ 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...).
- * ext/alsa/Makefile.am:
- * ext/alsa/gstalsa.c: (gst_alsa_init), (gst_alsa_get_caps),
- (gst_alsa_fixate_to_mimetype), (gst_alsa_fixate_field_nearest_int),
- (gst_alsa_link), (gst_alsa_close_audio):
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/alsa/gstalsasink.c: (gst_alsasink_get_type),
- (gst_alsasink_dispose), (gst_alsasink_base_init),
- (gst_alsasink_class_init), (gst_alsasink_init),
- (gst_alsasink_getcaps), (set_hwparams), (set_swparams),
- (alsasink_parse_spec), (gst_alsasink_open), (gst_alsasink_close),
- (xrun_recovery), (gst_alsasink_write), (gst_alsasink_delay),
- (gst_alsasink_reset):
- * ext/alsa/gstalsasink.h:
- Implement alsasink with simple open/write/close API.
- Make alsa dir build by disabling compilation of code.
+2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-28 Wim Taymans <wim@fluendo.com>
+ 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-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.
+2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-04-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- * gst/audioconvert/Makefile.am:
- * gst/audioscale/Makefile.am:
- Fix part of the build. Come on guys, autogen didn't even work :)
+2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.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().
- * configure.ac:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/media-info/.cvsignore:
- * gst-libs/gst/media-info/Makefile.am:
- * gst-libs/gst/media-info/README:
- * gst-libs/gst/media-info/media-info-priv.c:
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c:
- * gst-libs/gst/media-info/media-info.c:
- * gst-libs/gst/media-info/media-info.h:
- * gst-libs/gst/media-info/media-info.vcproj:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-media-info-uninstalled.pc.in:
- * pkgconfig/gstreamer-media-info.pc.in:
- Remove media-info, which is also successed by playbin (see Totem
- implementation).
+2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * configure.ac:
- * examples/Makefile.am:
- * examples/gstplay/.cvsignore:
- * examples/gstplay/Makefile.am:
- * examples/gstplay/player.c:
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/play/.cvsignore:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.c:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/play/play.vcproj:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-play.pc.in:
- Remove libgstplay, playbin is now the official successor.
+2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
+ Original commit message from CVS:
+ * ext/pango/Makefile.am:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
+ (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
+ Use gstvideo functions to calculate strides and plane offsets. Fixes
+ rendering issue ('ghost' images of the text on the chroma planes)
+ with widths or heights that are not multiples of 8 (#506659 and
+ probably also #485729).
+ * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
+ (main):
+ Test with odd height/width too.
-2005-04-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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).
+ 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.
-2005-04-25 David Schleef <ds@schleef.org>
+2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- * 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:
+ 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.
-2005-04-25 David Schleef <ds@schleef.org>
+2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
- 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:
+ * 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.
- Convert to 0.9 API, seems to work:
- * sys/ximage/Makefile.am:
- * sys/ximage/ximagesink.c:
+2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-04-24 David Schleef <ds@schleef.org>
+ 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.
- Link plugins against libraries:
- * ext/alsa/Makefile.am:
- * gst/tcp/Makefile.am:
+2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- Remove asm code that should be in liboil
- * gst/videoscale/Makefile.am:
- * gst/videoscale/videoscale_x86_asm.s:
+ 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.
- gettext wants these checked in:
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/it.po:
- * po/nb.po:
- * po/nl.po:
- * po/or.po:
- * po/sq.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- * po/vi.po:
+2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.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).
+
+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.
+
+2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c:
+ (gst_alsa_mixer_track_update_alsa_capabilities):
+ Make sure playback volumes aren't accidentally overwritten by
+ capture volumes if an alsa mixer track has both playback and
+ capture capabilities: we create two GstMixerTracks in that
+ case, so make sure we query only the alsa capabilities that
+ refer to the type of GstMixerTrack we created from the dual
+ capability alsa element. Should fix issues with Audigy2 sound
+ cards (#518082).
+
+2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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.
+
+2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.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.
+
+2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ 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.
+
+2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.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.
+
+2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/libs/video.c: More checks.
+ Original commit message from CVS:
+ * tests/check/libs/video.c:
+ More checks.
+
+2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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.
+
+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.
-2005-04-24 David Schleef <ds@schleef.org>
+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 ';'.
- Convert gst_main() to g_main_loop_run()
- * gst/playback/decodetest.c: (main):
- * gst/playback/test2.c: (main):
- * gst/playback/test3.c: (main):
- * gst/playback/test4.c: (main):
+2008-05-21 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.
- Link plugins against libraries:
- * ext/libvisual/Makefile.am:
- * sys/xvimage/Makefile.am:
+2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Fix logic in last commit.
+
+2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Passthrough the channel positions if the number of output channels is
+ the same as the number of input channels, the input had a channel
+ layout and downstream requests no special one. We did this already for
+ > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
+
+2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
+ (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_received_headers_callback),
+ (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ Set the ICY caps on the srcpad from where they get picked up by the base
+ class now and set on the outgoing buffers.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
+ BaseSrc now sets the caps on outgoing buffers automatically.
+
+2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_resample_slaving),
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_change_state):
+ Change the way in which the ringbuffer is started when dealing with a
+ slaved clock and latency. We now sync to the clock until we reach
+ upstream latency before starting the ringbuffer. This has the effect
+ that we can accurately align the master and slave clocks and let the
+ rate correction code take care of the initial drift or rounding errors
+ instead of leaving them uncorrected with the old approach.
+
+2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_fixate_channels):
+ Correctly set the default channel positions when converting to 8
+ channels.
+
+2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Error out if we don't have the required version of core.
+ Original commit message from CVS:
+ * configure.ac:
+ Error out if we don't have the required version of core.
-2005-04-24 David Schleef <ds@schleef.org>
-
- * configure.ac: Remove idct and resample libs
- * gst-libs/gst/Makefile.am: same
-
- Remove usage of gst_library_load():
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/libvisual/visual.c: (plugin_init):
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/theora/theora.c: (plugin_init):
- * ext/vorbis/vorbis.c: (plugin_init):
- * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
- * gst/audioscale/gstaudioscale.c:
- * gst/adder/gstadder.c: (plugin_init):
- * gst/audioconvert/plugin.c: (plugin_init):
- * sys/ximage/ximagesink.c: (plugin_init):
- * sys/xvimage/xvimagesink.c: (plugin_init):
- * gst/tcp/gsttcpplugin.c: (plugin_init):
-
- Link plugins against libraries:
- * ext/ogg/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/vorbis/Makefile.am:
- * gst/audioconvert/Makefile.am:
+2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.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: 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.
- 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.
+2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-04-20 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/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>
+2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c:
+ Adder doesn't support 24 bit samples so don't claim it supports them
+ in the pad template caps.
+
+2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain):
+ Validate the RTP packet before further processing it. It's just too
+ dangerous to accept random packets and people are not forced to use a
+ jitterbuffer or session manager to filter out the bad packets.
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_set_extension_data),
+ (gst_rtp_buffer_get_payload_subbuffer):
+ Small cleanups.
+ When setting extension data in a buffer that is too small, we fail and
+ we should not set the extension bit.
+ Change GST_WARNINGS into g_warning because they really are
+ programming errors.
+ * tests/check/libs/rtp.c: (GST_START_TEST):
+ Catch the g_warnings now in the unit tests and that fact that failing to
+ set extension data left the extension bit untouched.
+
+2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
+ Revert previous change which made basetransform handle buffer_alloc
+ and which breaks things badly in the non-passthrough case since it
+ returned buffers with a different (ie. sometimes smaller) size than
+ the size requested.
- * ext/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.
+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.
+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.
-2005-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2005-02-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- hunting season on 0.9 is now OPEN
+ gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
+ (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
+ (mpeg_video_stream_type_find):
+ Move scan helper thingy to the beginning of the file so we can use
+ it in other typefind functions. Rename it to something more
+ generic. Also improve handling of things towards the end of the
+ typefind data: peek as much as we can if we know the size of the
+ data, rather than just min_size.
-2005-02-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Kick the hell out of gcc for not warning me about a symbol conflict.
+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.
-2005-02-22 Luca Ognibene <luogni@tin.it>
+2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_link):
- Don't leak caps string (fixes #168134)
+2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init),
- (gst_jpegenc_init), (gst_jpegenc_finalize),
- (gst_jpegenc_change_state):
- Don't leak line buffers and context struct (fixes #168133).
+ gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/descriptions.c: (formats):
+ Fix description of video/x-flash-video.
-2005-02-21 Tim-Philipp Müller <tim at centricular dot net>
+2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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).
+ Remove some unused code.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
+ * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
+ Remove some unused code.
+ * gst/audioconvert/gstaudioquantize.c:
+ (gst_audio_quantize_free_noise_shaping):
+ Don't return before freeing the noise shaping history.
-2005-02-21 Maciej Katafiasz <mathrick@freedesktop.org>
+2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
+ Original commit message from CVS:
+ * tests/check/elements/subparse.c: (do_test),
+ (test_tmplayer_style3b), (subparse_suite):
+ Add unit test for the tmplayer variant from bug #530962.
-2005-02-21 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain):
- Make sure we only write to writable buffers
+2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-20 Tim-Philipp Müller <tim at centricular dot net>
+ 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/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Do actually fix invalid RIFF fmt header values for alaw
- and mulaw audio instead of just saying so.
+2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Give gst_riff_create_audio_caps_with_data() a chance to
- fix up broken format header fields before extracting any
- parameters from the header. (fixes #167633)
+ gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (feed_textbuf):
+ Fix detection of discontinuities based on the buffer offset (doesn't work
+ so well if no buffer offset is set) and also check for the DISCONT buffer
+ flag. This keeps the parser state from being reset after each buffer in
+ the unit test.
-2005-02-19 Martin Holters <martin.holters@gmx.de>
+2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ 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).
- * gst/audioconvert/bufferframesconvert.c:
- (buffer_frames_convert_link):
- Don't leak othercaps. (fixes #167878)
+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 :)
-2005-02-19 Arwed v. Merkatz <v.merkatz@gmx.net>
+2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
- * configure.ac:
- * ext/libvisual/visual.c: (gst_visual_srclink),
- (gst_visual_change_state):
- Support libvisual 0.2.0.
+ * 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.
-2005-02-18 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ 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.
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_chain):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_resync), (gst_jpegenc_chain):
- Use same rowstrides for I420 as used everywhere else.
+2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
-2005-02-17 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst/avi/gstavidemux.c: (gst_avi_demux_invert):
- Declare variables at beginning of block and make gcc-2.95 happy
- (fixes # 167482, patch by Gergely Nagy).
-
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- Move some includes into the header, so that struct sockaddr_in is
- defined when it should be defined on FreeBSD as well (fixes
- #167483).
-
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
- Don't pass uninitialised values to setsockopt() here either.
+2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-17 Luca Ognibene <luogni at tin dot it>
+ 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.
- Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+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/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send):
- Don't pass uninitialised values to setsockopt(). (fixes #167704)
+2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-16 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * gst/playback/gstplaybin.c: (add_sink):
- Invert bin_add/link order to workaround deadlock in opt.
+2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.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/modplug/gstmodplug.cc:
- Add missing break causing position queries to fail.
+2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-15 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate):
- Granpos can apparently be -1, which screws up calculations...
+2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
-2005-02-16 Jan Schmidt <thaytan@mad.scientist.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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_send_pending_navigation),
- (gst_ximagesink_navigation_send_event), (gst_ximagesink_finalize),
- (gst_ximagesink_init):
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_send_pending_navigation),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_finalize), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h:
- Use a mutex protected list to marshal navigation
- events into the stream thread from whichever thread
- sends them.
-
-2005-02-15 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/speed/demo-mp3.c: (time_tick_cb), (main):
- Display current position and track length; misc. clean-ups.
-
- * gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query),
- (speed_init), (speed_chain):
- Add query function, so that the stream length and current position
- get adjusted when queried (note that current position queries may
- still be wrong if the audio sink returns values based on buffer
- timestamps instead of passing on the query).
-
-2005-02-13 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
- (gst_audio_convert_channels):
- create channel conversion matrix when linking
- * gst/audioconvert/.cvsignore:
- * gst/audioconvert/Makefile.am:
- * gst/audioconvert/channelmixtest.c: (main):
- add (ugly) test that ensures stereo <=> mono conversion works
- correctly
+2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
-2005-02-13 Benjamin Otte <otte@gnome.org>
+ sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c:
+ Fix typo that causes the overlay keying color to bright green
+ on a 16-bit display. Dark grey good. Bright green bad.
- * gst/audioconvert/gstchannelmix.h:
- include missing header file
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_compatible):
- use same sign for both channels when converting to/from compatible
- channel. Previously used different signs made the signals cancel
- each other out and appear like silence. (fixes #167269)
+2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-12 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/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c:
- Convert to and from YV12 (fixes #156379).
-
-2005-02-12 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_sink_link), (gst_ximagesink_change_state),
- (gst_ximagesink_chain), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_expose), (gst_ximagesink_set_property),
- (gst_ximagesink_finalize), (gst_ximagesink_init): Protect interface
- methods from chain and negotiation and vice versa (Fixes #166142).
- * sys/ximage/ximagesink.h: Add stream_lock.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose): Check for xcontext before trying to link.
-
-2005-02-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_open):
- Don't send "Hey! You gave me a NULL pointer you naughty person" as
- error message when we can't open the DVD device (when dvdnav_open()
- fails, src->dvdnav is NULL, so dvdnav_err_to_string() will return
- the above). Send something more useful instead (fixes #167117).
-
-2005-02-11 Julien MOUTTE <julien@moutte.net>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
- (gst_xvimagesink_sink_link), (gst_xvimagesink_change_state),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_expose), (gst_xvimagesink_set_property),
- (gst_xvimagesink_finalize), (gst_xvimagesink_init): Protect interface
- methods from chain and negotiation and vice versa (Fixes #166142).
- Fix a possible bug of images in the buffer pool being discarded because
- we are looking at the wrong geometry.
- * sys/xvimage/xvimagesink.h: Add stream_lock.
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (crop_buffer): Change uint to
- unsigned int. (fixes #167128)
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * gst/librfb/Makefile.am: Testing stuff before committing is
- for wimps... and people with fast machines. Fix stupid
- mistake.
-
-2005-02-11 David Schleef <ds@schleef.org>
-
- * configure.ac: Pull in librfb from my CVS tree, because it is
- too small and annoying to be separate. Move rfbsrc plugin
- to gst/.
- * ext/Makefile.am:
- * ext/librfb/Makefile.am:
- * ext/librfb/gstrfbsrc.c:
- * gst/librfb/Makefile.am:
- * gst/librfb/gstrfbsrc.c:
- * gst/librfb/rfb.c:
- * gst/librfb/rfb.h:
- * gst/librfb/rfbbuffer.c:
- * gst/librfb/rfbbuffer.h:
- * gst/librfb/rfbbytestream.c:
- * gst/librfb/rfbbytestream.h:
- * gst/librfb/rfbcontext.h:
- * gst/librfb/rfbdecoder.c:
- * gst/librfb/rfbdecoder.h:
- * gst/librfb/rfbutil.h:
-
-2005-02-10 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst/speed/Makefile.am:
- * gst/speed/demo-mp3.c: (main):
- * gst/speed/filter.func:
- * gst/speed/gstspeed.c: (speed_link), (speed_parse_caps),
- (speed_class_init), (speed_init), (speed_chain_int16),
- (speed_chain_float32), (speed_chain), (speed_set_property),
- (speed_get_property), (speed_change_state):
- * gst/speed/gstspeed.h:
- Fix speed element and make it chain-based (fixes #156467),
- and make it handle more than one channel.
-
-2005-02-10 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dts/gstdtsdec.c: (gst_dtsdec_init), (gst_dtsdec_channels),
- (gst_dtsdec_handle_event), (gst_dtsdec_handle_frame),
- (gst_dtsdec_chain), (gst_dtsdec_change_state):
- * ext/dts/gstdtsdec.h:
- Don't clobber the stack constructing the channels array.
- Make the element chain-based. DTS tracks can now be played.
-
-2005-02-09 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-20 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.
- * 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.
+2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-09 David Schleef <ds@schleef.org>
+ 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().
- * testsuite/gst-lint: Check for non-statically scoped
- parent_class variables. This won't be a problem once
- plugins are loaded with RTLD_LOCAL.
+2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/mplex/gstmplexibitstream.cc:
- gcc madness.
+2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
-2005-02-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
+ Don't validate the payload if there isn't any.
+ Fixes #525915
- * ext/ogg/gstogmparse.c:
- * gst/debug/gstnavigationtest.c:
- Die, thou faulty symbol pollutors (non-static parent_class).
+2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
+ Use g_atomic_int_set() instead of gst_atomic_int_set().
- * ext/mplex/gstmplexibitstream.cc:
- Fix event handling (#165525).
+2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Return NULL instead of a gchar * array with one NULL element if we
+ don't get any supported URI schemes from GIO.
- * ext/mikmod/gstmikmod.c:
- * gst/modplug/gstmodplug.cc:
- Add missing endianness to template (fixes #165509).
+2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ Remove cpp style commented old code.
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_data):
- Fix wrong order of reading of optional bytes (#165290).
+2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstdecodebin2.c: Fix signal docs.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c:
+ Fix signal docs.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Implement FILLER event awareness.
+2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
+ (gst_text_overlay_init):
+ Fix textoverlay unit test again by making the supposed default
+ value for the wait-text property the actual default value.
+ Also fix Since: tag for new property.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_convert):
- Fix track calculations (#166208).
+2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-02-08 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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>
- * ext/libpng/gstpngdec.c: (gst_pngdec_init), (gst_pngdec_chain):
- * ext/libpng/gstpngenc.c:
- Fix byte-order, use proper fixed caps. Fixes #164197.
+ * 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.
-2005-02-08 Jan Schmidt <thaytan@mad.scientist.com>
+2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- Add dvdlpcmdec
+ * 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.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_reset),
- (free_all_buffers), (gst_mpeg2dec_alloc_buffer):
- Don't push buffers if the src pad isn't negotiated yet.
-
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format):
- Add support for 24-bit width.
-
- * gst/dvdlpcmdec/.cvsignore:
- * gst/dvdlpcmdec/Makefile.am:
- * gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_get_type),
- (gst_dvdlpcmdec_base_init), (gst_dvdlpcmdec_class_init),
- (gst_dvdlpcm_reset), (gst_dvdlpcmdec_init), (gst_dvdlpcmdec_link),
- (gst_dvdlpcmdec_chain), (gst_dvdlpcmdec_change_state),
- (plugin_init):
- * gst/dvdlpcmdec/gstdvdlpcmdec.h:
- New decoder for rearranging DVD LPCM into our audio/x-raw-int
- format. Needs support for the channels maps if someone can find
- a DVD LPCM track with > 2 channels.
-
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_handle_dvd_event),
- (gst_dvd_demux_send_discont), (gst_dvd_demux_handle_discont),
- (gst_dvd_demux_get_audio_stream), (gst_dvd_demux_process_private):
- * gst/mpegstream/gstdvddemux.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
- (gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_init_stream),
- (gst_mpeg_demux_send_subbuffer), (gst_mpeg_demux_handle_src_query):
- * gst/mpegstream/gstmpegdemux.h:
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_reset),
- (gst_mpeg_parse_parse_packhead), (gst_mpeg_parse_loop),
- (gst_mpeg_parse_get_rate), (gst_mpeg_parse_convert_src),
- (gst_mpeg_parse_handle_src_query),
- (gst_mpeg_parse_handle_src_event):
- Use audio/x-dvd-lpcm for LPCM output.
- Add DTS output.
-
-2005-02-08 Gergely Nagy <algernon@bonehunter.rulez.org>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/alpha/gstalphacolor.c: (gst_alpha_color_sink_link),
- (transform_rgb), (transform_bgr), (gst_alpha_color_chain):
- Add BGRA handling (#165736).
-
-2005-02-08 Francis Labonte <francis_labonte@hotmail.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/law/alaw-decode.c: (alawdec_link):
- * gst/law/alaw-encode.c: (alawenc_link):
- * gst/law/mulaw-decode.c: (mulawdec_link):
- * gst/law/mulaw-encode.c: (mulawenc_link):
- Fix caps memleaks (#166600).
-
-2005-02-08 Tim-Philipp Müller <tim at centricular dot net>
-
- * ext/tarkin/mem.h:
- * ext/tarkin/wavelet.h:
- * ext/tarkin/yuv.h:
- * gst/ffmpegcolorspace/avcodec.h:
- Include "_stdint.h" instead of <stdint.h>. Fixes build on
- systems that don't have stdint.h, like Solaris9 (fixes #166631).
-
-2005-02-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_clear),
- (gst_xvimagesink_change_state):
- Clear window on PAUSED->READY instead of READY->PAUSED. Stop
- Xv video (and thereby regenerate Xv colourkey) in clear() so
- that PLAY -> READY -> PLAY works (fixes #162504).
-
-2005-02-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_getcaps):
- Switch to list instead of range, since MJPEG-devices really just
- support decimations, not any size.
-
-2005-02-05 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_open_decoder),
- (gst_mpeg2dec_reset), (free_all_buffers),
- (gst_mpeg2dec_alloc_buffer), (handle_sequence):
- * ext/mpeg2dec/gstmpeg2dec.h:
- The libmpeg2 user-allocated buffer management is awkward,
- to say the least. Hopefully this fixes things.
-
-2005-02-04 Andy Wingo <wingo@pobox.com>
-
- * gst/audioconvert/bufferframesconvert.c
- (buffer_frames_convert_fixate): New function, fixates to 256
- frames per buffer by default. (Much better than 1.)
- (buffer_frames_convert_init): Set the fixate function for both src
- and sink pad.
- (buffer_frames_convert_link): After success setting nonfixed caps,
- get the negotiated caps so we can know how many buffer-frames it
- will be. No idea how this worked at all before.
-
-2005-02-05 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_init),
- (gst_mpeg2dec_close_decoder), (put_buffer), (check_buffer),
- (free_buffer), (free_all_buffers), (gst_mpeg2dec_alloc_buffer),
- (handle_sequence), (handle_picture):
- * ext/mpeg2dec/gstmpeg2dec.h:
- Rearrange buffer tracking and refcounting and refactor
- a little for readability.
-
-2005-02-04 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_get_type),
- (gst_v4ljpegsrc_base_init), (gst_v4ljpegsrc_class_init),
- (gst_v4ljpegsrc_init), (gst_v4ljpegsrc_src_link),
- (gst_v4ljpegsrc_getcaps), (gst_v4ljpegsrc_get):
- * sys/v4l/gstv4ljpegsrc.h:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_open), (gst_v4lsrc_src_link):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_palette_name),
- (gst_v4lsrc_get_fps):
- * sys/v4l/v4lsrc_calls.h:
- Add new v4ljpegsrc for handling the ov51x hacky "I'll give
- you jpeg inside rgb frames" driver.
- Don't error in the v4lsrc link function, just return
- REFUSED.
-
-2005-02-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
- (gst_qcamsrc_open):
- Use GST_ELEMENT_ERROR, not g_warning, if open failed.
-
-2005-02-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
- Change caps on MJPEG-B so it doesn't interfere with MJPEG/JPEG.
-
-2005-02-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_change_state):
- Reset negotiated state on PAUSED->READY.
-
-2005-02-02 David Schleef <ds@schleef.org>
-
- * configure.ac: Put DEFAULT_AUDIOSINK in config.h and use
- whereever possible. (Fixes #165997)
- * examples/capsfilter/capsfilter1.c: (main):
- * examples/dynparams/filter.c: (create_ui):
- * examples/seeking/cdparanoia.c: (get_track_info), (main):
- * examples/seeking/chained.c: (main):
- * examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
- (make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
- (make_vorbis_pipeline), (make_mp3_pipeline), (make_avi_pipeline),
- (make_mpeg_pipeline), (make_mpegnt_pipeline):
- * examples/seeking/spider_seek.c: (make_spider_pipeline):
- * examples/switch/switcher.c: (main):
- * ext/dv/demo-play.c: (main):
- * ext/faad/gstfaad.c: (gst_faad_change_state):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/smoothwave/demo-osssrc.c: (main):
- * gst-libs/gst/gconf/gconf.c: (gst_gconf_set_string),
- (gst_gconf_render_bin_from_description),
- (gst_gconf_get_default_audio_sink),
- (gst_gconf_get_default_video_sink),
- (gst_gconf_get_default_audio_src),
- (gst_gconf_get_default_video_src),
- (gst_gconf_get_default_visualization_element):
- * gst/level/demo.c: (main):
- * gst/level/plot.c: (main):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element):
- * gst/playback/test.c: (gen_video_element), (gen_audio_element):
- * gst/playondemand/demo-mp3.c: (setup_pipeline):
- * gst/sine/demo-dparams.c: (main):
- * gst/spectrum/demo-osssrc.c: (main):
- * gst/speed/demo-mp3.c: (main):
- * gst/volume/demo.c: (main):
- * testsuite/embed/embed.c: (main):
-
-2005-02-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_class_init),
- (gst_tcpclientsink_finalize):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_class_init),
- (gst_tcpclientsrc_finalize):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_init), (gst_tcpserversink_finalize):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_class_init),
- (gst_tcpserversrc_init), (gst_tcpserversrc_finalize):
- Don't leak the hostname when shutting down.
- In tcpserversrc, take a copy of the default hostname.
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_iso_receive):
- Set caps to systemstream=TRUE.
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * testsuite/Makefile.am:
- Fix more OSX buildbots.
-
-2005-02-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c:
- Don't send things to NULL PAD_PEERs
-
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_chain):
- Copy-on-write the incoming buffer.
-
- * gst/mpegstream/gstdvddemux.h:
- * gst/mpegstream/gstmpegclock.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (normal_seek), (gst_mpeg_demux_handle_src_event):
- * gst/mpegstream/gstmpegdemux.h:
- * gst/mpegstream/gstmpegpacketize.h:
- * gst/mpegstream/gstmpegparse.c:
- (gst_mpeg_parse_update_streaminfo), (gst_mpeg_parse_reset),
- (gst_mpeg_parse_handle_discont), (gst_mpeg_parse_parse_packhead),
- (gst_mpeg_parse_loop), (gst_mpeg_parse_get_rate),
- (gst_mpeg_parse_convert_src), (gst_mpeg_parse_handle_src_query),
- (gst_mpeg_parse_handle_src_event), (gst_mpeg_parse_change_state):
- * gst/mpegstream/gstmpegparse.h:
- * gst/mpegstream/gstrfc2250enc.h:
- Various changes to the way time is computed that make seeking and
- total time estimation much better here.
- Use G_BEGIN/END_DECLS instead of __cplusplus
-
- * gst/videocrop/gstvideocrop.c: (gst_video_crop_chain):
- Use gst_buffer_stamp instead of only copying the TIMESTAMP
-
-2005-02-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/subparse/gstsubparse.c:
- Fix OSX buildbot.
+ Merge other changes from 0.10.19 release branch.
+ Original commit message from CVS:
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-base.doap:
+ Merge other changes from 0.10.19 release branch.
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_enc_chain), (theora_enc_change_state):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_chain),
- (gst_vorbisenc_change_state):
- * ext/vorbis/vorbisenc.h:
- Set granulepos and timestamp correctly for streams not
- starting at 0, taking into account the initial delay.
+ gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ * gst/playback/gstplayback.c: (plugin_init):
+ * gst/volume/gstvolume.c: (plugin_init):
+ Work around missing bits of thread-safety on older GLibs some
+ more to avoid assertions when starting up multiple playbin
+ objects concurrently (see #512382).
-2005-01-31 Tim-Philipp Müller <tim at centricular dot net>
+2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstdvddemux.c:
- Add audio/x-dts to audio pad template caps
+ gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
+ Original commit message from CVS:
+ * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
+ Remove some more fields.
-2005-01-30 David Schleef <ds@schleef.org>
+2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
- * 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.
+ configure.ac: Actually build dlls when cross-compiling with mingw32.
+ Original commit message from CVS:
+ Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
+ * configure.ac:
+ Actually build dlls when cross-compiling with mingw32.
+ Fixes bug #526247.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
- Add 3IV2 fourcc.
+ configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ tests/examples/seek/seek.c: Add statusbar.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
+ (msg_buffering), (connect_bus_signals), (main):
+ Add statusbar.
+ Add buffering support with feedback in the statusbar.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- D'oh, reference the palette data, not the palette structure.
- Fixes color distortion in #132341.
+ ext/ogg/gstoggmux.c: Fix sample pipeline description.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Fix sample pipeline description.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- PAR can be non-fixed when not provided as argument (#162626).
+ docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-overrides.txt:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Update introspection data.
+ * ext/ogg/gstoggmux.c:
+ Document oggmux.
+ * gst/playback/gstdecodebin2.c:
+ Don't use gtk-doc style comment start for private stuff, but make it
+ formatted like this for consistency.
-2005-01-29 David Moore <dcm@acm.org>
+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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header):
- Re-apply patch from #142272 that allows non-seekable sources,
- re-proposed by Daniel Drake <dsd@gentoo.org>.
+ 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.
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_init):
- Use the src template for creating the src pad (#162330).
+2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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>
+2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/Makefile.am:
- Unbreak buildbot.
+ 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.
-2005-01-28 Andy Wingo <wingo@pobox.com>
+2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dv/gstdvdec.c: Change the pixel aspect ratio of dvdec output
- to reflect a different dubious internet source. Add a reference
- and some commentary.
+ 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.
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_class_init):
+ * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_class_init):
+ * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
+ * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
+ * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
+ * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
+ * ext/pango/gsttextrender.c: (gst_text_render_class_init):
+ * ext/theora/theoradec.c: (gst_theora_dec_class_init):
+ * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
+ * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_class_init):
+ * gst-libs/gst/interfaces/mixertrack.c:
+ (gst_mixer_track_class_init):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init):
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_class_init):
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
+ * gst/audioresample/gstaudioresample.c:
+ (gst_audioresample_class_init):
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init):
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (preroll_unlinked):
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
+ * gst/playback/gstqueue2.c: (gst_queue_class_init):
+ * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
+ * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
+ (gst_stream_selector_class_init):
+ * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
+ * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
+ * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
+ * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init):
+ * gst/volume/gstvolume.c: (gst_volume_class_init):
+ * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
+ * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
+ * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
+ Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
+ static strings (i.e. all). This gives us less memory usage,
+ fewer allocations and thus less memory defragmentation. Depend
+ on core CVS for this. Fixes bug #523806.
+
+2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Filter http and https protocols. GIO/GVfs handles them but it's
+ impossible to implement iradio/icecast with it. Better use
+ souphttpsrc or something else for this.
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
+ If getting the file informations by a query fails try it with the
+ seek-to-end trick too.
+
+2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (gst_volume_interface_supported),
+ (gst_volume_base_init), (gst_volume_class_init),
+ (volume_process_double), (volume_process_float),
+ (volume_transform_ip), (plugin_init):
+ memset buffers to zero if we get a GAP buffer. We usually see a
+ buffer as one unit so let's handle it as one and don't care about
+ volume changes while processing one buffer.
+ Also clean up some stuff a bit.
+
+2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
+ (gst_audio_convert_create_silence_buffer),
+ (gst_audio_convert_transform):
+ Make audioconvert GAP-aware by outputting silence buffers when the
+ input has the GAP flag set. This is up to 8x faster.
+ Based on a patch by Stefan Kost. Fixes bug #517813.
+
+2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_process_double):
+ Use oil_scalarmultiply_f64_ns() for double processing when it's
+ available at compile time.
+
+2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
+ Original commit message from CVS:
+ * configure.ac:
+ Fix lrint/lrintf checks to actually work. These functions are
+ in libm on Linux at least so try to link to it.
+
+2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ configure.ac: Back to development - 0.10.18.1
+ Original commit message from CVS:
+ * configure.ac:
+ Back to development - 0.10.18.1
- * gst/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.
+=== release 0.10.18 ===
-2005-01-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* 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>
-
+ * 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:
- * 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-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.
-
-2005-01-25 Edward Hervey <bilboed@bilboed.com>
-
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/effectv/gstquark.c: (gst_quarktv_class_init),
- (gst_quarktv_change_state), (gst_quarktv_dispose):
- Memory free'ing location fix (#164708).
-
-2005-01-25 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
-
-2005-01-25 Andy Wingo <wingo@pobox.com>
-
- * sys/v4l/gstv4lelement.c (gst_v4l_iface_supported): Fix compile
- for #ifndef HAVE_XVIDEO.
-
-2005-01-24 Jeffrey C. Ollie
-
- reviewed by: Maciej Katafiasz <mathrick@freedesktop.org>
-
- * 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.
-
-2005-01-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Audio can be <8000Hz.
-
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * win32/common/config.h:
+ Release 0.10.18
+ Original commit message from CVS:
+ Release 0.10.18
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Explicit state change to workaround refcount bugs.
+2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-22 Ronald S. Bultje <rbultje@ronald.bitfreak.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/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.
- * gst/avi/gstavimux.c: (gst_avimux_write_tag),
- (gst_avimux_riff_get_avi_header):
- Fix...
+2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
- (gst_riff_read_element_data):
- * gst-libs/gst/riff/riff-read.h:
- Add _peek version (req'ed in CDXA).
- * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
- (gst_cdxaparse_loop):
- Fix parsing in playbin.
- * gst/playback/gstdecodebin.c: (close_pad_link):
- Ignore current_ pads, they cause major annoyance.
+2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Safety guard.
+2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/avi/gstavimux.c: (gst_avimux_write_tag):
- Fix padding...
+2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.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/matroska/ebml-read.c: (gst_ebml_read_buffer):
- Allow for 0-sized buffers. Fixes length query problems in
- starwars.mkv from the testsuite.
+2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
-2005-01-19 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * gst/videobox/gstvideobox.c: (gst_video_box_copy_plane_i420),
- (gst_video_box_i420), (gst_video_box_chain):
- Fix row strides for I420 (fixes #163159)
-
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-11 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.
- * 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.
+2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
+ (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
+ (gst_mixer_message_get_type),
+ (gst_mixer_message_parse_option_changed),
+ (gst_mixer_message_parse_options_list_changed):
+ * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
+ (GST_MIXER_MESSAGE_OPTION_CHANGED),
+ (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
+ (GST_MIXER_MESSAGE_MIXER_CHANGED):
+ API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
+ and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add intel-h263.
+2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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)
- * 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.
+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).
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
- Elements may already be destroyed when this function is called.
+ 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).
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (gst_qtdemux_handle_esds):
- More memory leak fixes (#149162).
+ 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.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_add_stream):
- Fix two memleaks.
+ 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.
-2005-01-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
- Argh...
+ 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.
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * 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.
+2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-17 Tim-Philipp Müller <tim at centricular dot 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).
- * ext/dvdread/dvdreadsrc.c: (get_next_cell_for):
- Fix compile warnings on Solaris 10 buildbot
+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>
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+ * 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/dvdread/dvdreadsrc.c: (_read):
- Don't read beyond the last cell in a chapter (fixes
- invalid memory access)
+2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-17 Tim-Philipp Müller <tim at centricular dot net>
+ 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.
- * 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)
+2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps):
- Fix BGRA32 caps (#164209).
+ gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
+ (plugin_init):
+ Add midi typefinder, copied from the timidity plugin.
-2005-01-17 Gergely Nagy <algernon@bonehunter.rulez.org>
+2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- alpha_mask can be RGBA/ABGR. Fixes #164265.
+2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-17 Francis Labonte <francis_labonte@hotmail.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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
- * 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).
+ configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
+ Original commit message from CVS:
+ Patch by: Brian Cameron <brian dot cameron at sun dot com>
+ * configure.ac:
+ Don't hardcode -Wall and -Werror for configure checks, this fails
+ with non-GCC compilers. Fixes bug #517991.
-2005-01-17 Steve Baker <steve@stevebaker.org>
+2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.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/typefind/gsttypefindfunctions.c: (aiff_type_find),
- (svx_type_find), (sds_type_find), (ircam_type_find), (plugin_init):
- Add libsndfile typefind functions (#163309).
+2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * tools/gst-launch-ext-m.m:
- Add .aac, fix .m1v/.m2v (#163891).
+2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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'.
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Of course, only pause if group is done...
+2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/playback/gstplaybasebin.c: (probe_triggered):
- Thread safety.
+2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-16 Jan Schmidt <thaytan@mad.scientist.com>
+ 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.
- * ext/swfdec/gstswfdec.c: (gst_swfdec_change_state):
- Don't return state change success when the parent
- failed.
+2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
-2005-01-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/avi/gstavimux.c: (gst_avimux_handle_event):
- Free events (fix memleak in #162905).
+2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-15 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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).
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Add screenshot conversion code from totem.
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
+ (create_element), (gst_play_frame_conv_convert):
+ * gst/playback/gstscreenshot.h:
+ Add screenshot conversion code from totem.
+ * gst/playback/gstplay-marshal.list:
+ * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
+ (gst_play_bin_class_init), (gst_play_bin_convert_frame),
+ (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
+ Implement frame property to get a color-unconverted snapshot.
+ Implement convert-frame action signal to get a converted snapshot image.
+ Configure connection speed in uridecodebin.
+ Document some more properties.
+ * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
+ (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_get_last_frame):
+ * gst/playback/gstplaysink.h:
+ Use last-buffer property of the video sink to get a video snapshot.
+ * tests/examples/seek/seek.c: (shot_cb), (main):
+ Add snapshot button for playbin2 and use the frame property to save the
+ frame as a png in the current directory.
+
+2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
+ Original commit message from CVS:
+ Patch by: Josep Torra Valles <josep at fluendo dot com>
+ * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
+ (plugin_init):
+ Add typefinding support for h264 elementary streams.
+ Fixes bug #517420.
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_caps_to_pixfmt):
- Fix for depth = 15. Fixes #161675.
+2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2005-01-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Require CVS of core for new API in collectpads.
+ Original commit message from CVS:
+ * configure.ac:
+ Require CVS of core for new API in collectpads.
+ * gst/adder/gstadder.c:
+ Use new API to make adder sparse stream aware.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Set FPS correctly, even for webcams and the like.
- * sys/v4l/v4l_calls.c: (gst_v4l_set_chan_norm):
- Don error on setting while capturing.
+2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
+ (no_more_pads_cb):
+ Get the object data correct so that we can remove our channels
+ correctly.
+ * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
+ (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_request_pad):
+ Add option to disable async behaviour in the sinks when possible. This
+ makes it possible to avoid an audio queue when dealing with
+ visualisations.
+ Add option to add a queue for the audio path.
+ * tests/examples/seek/seek.c: (clear_streams), (update_streams),
+ (main):
+ Disable the vis checkbox to match the defaults of playbin2.
+ Only get the stream info when we need to.
+
+2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Don't use async operations as they require a running main loop.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
+ (gst_gio_base_sink_set_stream):
+ * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
+ (gst_gio_base_src_set_stream):
+ * ext/gio/gstgiosink.c: (gst_gio_sink_start):
+ * ext/gio/gstgiosrc.c: (gst_gio_src_start):
+ Don't use async operations as they require a running main loop.
+ This makes us block again when closing streams and unable
+ to mount the enclosing volume of an URI if it isn't yet.
+
+2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
+ Original commit message from CVS:
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
+ (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
+ (gen_vis_chain), (gst_play_sink_reconfigure),
+ (gst_play_sink_request_pad):
+ Move tee in front of the audio and vis pipelines.
+ Add queue for audio for now.
+ Add visualisation support.
+ * tests/examples/seek/seek.c: (main):
+ Visualisation is by default disabled.
+
+2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Improve debugging a bit.
+ Original commit message from CVS:
+ * ext/gio/gstgiobasesink.c: (close_stream_cb):
+ * ext/gio/gstgiobasesrc.c: (close_stream_cb):
+ Improve debugging a bit.
+ * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
+ * ext/gio/gstgiosink.h:
+ * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
+ * ext/gio/gstgiosrc.h:
+ Try to mount the enclosing volume of a GFile if it isn't mounted
+ yet. This requires us to wait for an async operation to finish, done
+ with an nested GMainLoop. Authentication is not supported yet, will
+ come later.
+
+2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Add mute property.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (gst_play_bin_set_property), (gst_play_bin_get_property),
+ (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
+ * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
+ (gst_play_sink_get_mute), (gen_audio_chain):
+ * gst/playback/gstplaysink.h:
+ Add mute property.
+ * gst/playback/gststreamselector.c: (gst_selector_pad_event),
+ (gst_selector_pad_chain):
+ * gst/playback/gststreamselector.h:
+ Make sure we forward the event only once.
+ * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
+ Add and implement the mute button for playbin2.
+
+2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
+
+ ext/alsa/gstalsasink.c: Add some more debug info.
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
+ Add some more debug info.
+ Make sure we never return a negative delay. Fixes #516246.
+
+2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
+ Revert patch that makes the sink hold the object lock when
+ calling snd_pcm_delay(), since it breaks playback for me.
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: Add some seek flags when changing rate.
+ Original commit message from CVS:
+ 2008-02-12 Julien Moutte <julien@fluendo.com>
+ * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
+ some seek flags when changing rate.
+
+2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer):
+ Fix potential leaks.
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
+ Fix leak when there is no function configured.
+
+2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
+ Original commit message from CVS:
+ * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
+ (gst_v4lsrc_buffer_finalize):
+ Correctly chain up the finalize method.
+
+2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
+ Original commit message from CVS:
+ * ext/gio/gstgiostreamsink.c:
+ * ext/gio/gstgiostreamsrc.c:
+ Add documentation and example code for giostreamsink/giostreamsrc.
+ * tests/check/pipelines/gio.c: (GST_START_TEST):
+ Ask the GMemoryOutputStream for the data instead of assuming that
+ the pointer to the data stayed the same. It could've been realloc'ed.
+
+2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
+ Original commit message from CVS:
+ * ext/gio/gstgiosink.c:
+ * ext/gio/gstgiosrc.c:
+ Make the documentation of giosink/giosrc complete, large parts
+ are based on the gnomevfssink/gnomevfssrc docs.
+
+2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ docs/plugins/: Add the GIO documentation again and while at that run make update.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-queue2.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-uridecodebin.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ Add the GIO documentation again and while at that run make update.
+
+2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be...
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
+ * ext/alsa/gstalsasink.c: (set_swparams):
+ * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
+ Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
+ against libasound >= 1.0.16, since it's been deprecated in
+ 0.10.16, and alignment is always 1 then, apparently. (#512899)
- * ext/dv/gstdvdec.c:
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
- I'm a bad boy. using /1001. to force C to do float division
- and not integer division (as it did in my last commit)
- Thanks to David I. Lehn for pointing this mistake.
+2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_audio_element):
+ * gst/playback/gstplaysink.c: (gen_audio_chain):
+ Handle case where we can't create the volume element a bit
+ better (#514307).
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Revert Johan's 1.35->1.36 since it breaks compat.
+2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-14 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ ext/gnomevfs/: Add support for https protocol. Fixes #510229.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
+ * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
+ Add support for https protocol. Fixes #510229.
- * ext/dv/gstdvdec.c:
- * ext/libfame/gstlibfame.c:
- * gst/subparse/gstsubparse.c: (parse_mdvdsub):
- * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect):
- replace framerate aproximations by their real value
- (24000/1001, 30000/1001, 60000/1001)
- Finish fixing bug #164049
+2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
-2005-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * ext/alsa/gstalsa.c: (gst_alsa_rates_probe):
- Fix for if items are already in list...
+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.
-2005-01-12 Benjamin Otte <otte@gnome.org>
+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/adder/gstadder.c: (gst_adder_loop):
- fix adder a bit so it doesn't screw up with events as much anymore
+2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
-2005-01-12 Jan Schmidt <thaytan@mad.scientist.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
- * 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.
+2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/avi/gstavimux.c: (gst_avimux_class_init),
- (gst_avimux_pad_unlink), (gst_avimux_release_pad):
- Reusability fixes.
+2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/gstalsa.c: (gst_alsa_rates_probe):
- Fix dmix.
+2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * 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.
+2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-11 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * 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.
+2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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).
- * 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
+2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix the tests environment's GST_PLUGIN_PATH: we want the directory
+ with the core's plugins first and our local build directories last,
+ since we might be building against an installed core, and that
+ core's plugin directory may contain older or other versions of
+ our own -base plugins, but we really do want to test our local
+ ones (if there are multiple plugins or element factories with the
+ same name, those inspected last will trump those read in earlier).
+ Fixes #512740 for the most part.
- * ext/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.
+2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Use gmtime_r if available as gmtime is not MT-safe.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Use gmtime_r if available as gmtime is not MT-safe.
+ Fixes bug #511810.
+
+2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Cast glong to time_t as time_t might have a different type on
+ other platforms, like FreeBSD, and we get a compiler warning
+ otherwise. Fixes bug #511825.
+
+2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
+ (get_group), (get_n_pads), (gst_play_bin_get_property),
+ (pad_added_cb), (no_more_pads_cb), (perform_eos),
+ (autoplug_select_cb), (deactivate_group):
+ Remove stream-info, we going for something easier.
+ Refactor getting the current group.
+ Implement getting the number of audio/video/text streams.
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_class_init), (gst_stream_selector_init),
+ (gst_stream_selector_get_property),
+ (gst_stream_selector_request_new_pad),
+ (gst_stream_selector_release_pad):
+ * gst/playback/gststreamselector.h:
+ Add property for number of pads.
+ * tests/examples/seek/seek.c: (set_scale), (update_flag),
+ (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
+ (text_toggle_cb), (update_streams), (msg_async_done),
+ (msg_state_changed), (main):
+ Block slider callback when updating the slider position.
+ Add gui elements for controlling playbin2.
+ Add callback for async_done that updates position/duration.
+
+2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/plugins/: First round of plugin docs cleansups.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.prerequisites:
+ First round of plugin docs cleansups.
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Regenerate.
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggmux.h:
+ Add header for oggmux. the c-file needs a doc blob still.
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+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.
- * 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
+2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Don't crash, biatch! :).
+=== release 0.10.17 ===
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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...
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * 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-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.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
- Also shove tags on kid pads.
+ gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
+ Original commit message from CVS:
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
+ Cast glong to time_t as time_t might have a different type on
+ other platforms, like FreeBSD, and we get a compiler warning
+ otherwise. Fixes bug #511825.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
- Don't bail on unknown events.
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
- Don't crash on events before negotiation.
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- Send tags on pads, too.
- * gst/playback/gststreamselector.c:
- (gst_stream_selector_request_new_pad):
- Forward events on first pad if no input was selected yet.
+ gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_class_init):
+ Initialize the GstRingerBuffer class to get it's debug category
+ initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
+ category and otherwise we get some g_critical(). Fixes bug #512334.
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
-2005-01-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+=== release 0.10.16 ===
- * sys/v4l/gstv4lxoverlay.c: (gst_v4l_xoverlay_interface_init),
- (gst_v4l_xoverlay_set_xwindow_id):
- * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_interface_init),
- (gst_v4l2_xoverlay_set_xwindow_id):
- Add debug categories, fix overlay disabling.
+2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2005-01-10 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ * 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
- * ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_get_caps):
- * ext/alsa/gstalsa.h:
- Add HW probing for period_count/size and buffer_size MIX/MAX
- Adjust default/user defined value if out of bounds
- Should fix bug #162024
+2008-01-28 22:15:47 +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/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.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event):
- Fix warning (#161191).
+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
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2008-01-10 17:55:53 +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)
+ gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ Ref audio clock class from a thread-safe context to make sure
+ we're not bit by GObjects lack of thread-safety here (#349410),
+ however unlikely that may be in practice.
-2005-01-09 Koop Mast <kwm@rainbow-runner.nl>
+2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/tta/gstttaparse.c: (gst_tta_src_event):
- Fix gcc-2.95 compile (#163485).
+2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.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).
- * configure.ac:
- * ext/flac/gstflacenc.c: (gst_flacenc_init),
- (gst_flacenc_seek_callback), (gst_flacenc_write_callback),
- (gst_flacenc_tell_callback), (gst_flacenc_chain),
- (gst_flacenc_change_state):
- * ext/flac/gstflacenc.h:
- Update for API change in flac-1.1.1. Update requirement in
- configure.ac. Fixes #162974.
+2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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):
- Remove hack to get rid of assert and get rid of unlinked
- signals properly.
+2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * testsuite/embed/Makefile.am:
- test->noinst, fix make test in buildbot.
+2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
-2005-01-09 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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).
- * 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.
+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.
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Fix hanging subs.
+2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * 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>.
+2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * testsuite/embed/Makefile.am:
- Fix buildbot.
+2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
-2005-01-09 Gergely Nagy <algernon@bonehunter.rulez.org>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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).
- * 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).
+2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
-2005-01-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.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-plugins.spec.in:
- Add subparse.
+2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * configure.ac:
- Since we use functions from CVS, up requirement.
+2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.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/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>
+2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_new):
- * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_get_type),
- (gst_ogm_text_parse_base_init), (gst_ogm_text_parse_init),
- (gst_ogm_parse_get_sink_querytypes), (gst_ogm_parse_sink_convert),
- (gst_ogm_parse_sink_query), (gst_ogm_parse_chain),
- (gst_ogm_parse_plugin_init):
- * ext/pango/gsttextoverlay.c: (gst_textoverlay_linkedpads),
- (gst_textoverlay_link), (gst_textoverlay_getcaps),
- (gst_textoverlay_event), (gst_textoverlay_video_chain),
- (gst_textoverlay_loop), (gst_textoverlay_init), (plugin_init):
- * ext/pango/gsttextoverlay.h:
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
- (gst_matroska_demux_handle_seek_event),
- (gst_matroska_demux_sync_streams),
- (gst_matroska_demux_parse_blockgroup),
- (gst_matroska_demux_subtitle_caps),
- (gst_matroska_demux_plugin_init):
- * gst/matroska/matroska-ids.h:
- * gst/playback/gstdecodebin.c: (close_pad_link):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (gen_preroll_element), (remove_groups), (add_stream),
- (new_decoded_pad), (setup_subtitles), (gen_source_element),
- (setup_source):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gen_text_element), (setup_sinks):
- * gst/subparse/Makefile.am:
- * gst/subparse/gstsubparse.c: (gst_subparse_get_type),
- (gst_subparse_base_init), (gst_subparse_class_init),
- (gst_subparse_init), (gst_subparse_formats),
- (gst_subparse_eventmask), (gst_subparse_event),
- (gst_subparse_handle_event), (convert_encoding), (get_next_line),
- (parse_mdvdsub), (parse_mdvdsub_init), (parse_subrip),
- (parse_subrip_deinit), (parse_subrip_init), (parse_mpsub),
- (parse_mpsub_deinit), (parse_mpsub_init),
- (gst_subparse_buffer_format_autodetect),
- (gst_subparse_format_autodetect), (gst_subparse_loop),
- (gst_subparse_change_state), (gst_subparse_type_find),
- (plugin_init):
- * gst/subparse/gstsubparse.h:
- * gst/typefind/gsttypefindfunctions.c: (ogmtext_type_find),
- (plugin_init):
- Add subtitle support, .sub parser (supports SRT and MPsub),
- OGM text support, Matroska UTF-8 text support, deadlock fixes
- all over the place, subtitle awareness in decodebin/playbin
- and some fixes to textoverlay to handle subtitles in a stream
- correctly. Fixes #100931.
-
-2005-01-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- Check for pad availability before doing a query on it.
-
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
-
- * ext/dv/gstdvdec.c:
- really fix bpp24/32 dvdec caps (classic rgba indeed)
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- don't send text tags if they are empty (bis repetita)
-
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
-
- * ext/dv/gstdvdec.c:
- remove unneeded comment from dvdec
- (related to DV 4CC codes in AVI files)
- moved them in gstreamer/docs/random/mimetypes
- * gst/asfdemux/gstasfdemux.c:
- (gst_asf_demux_process_ext_content_desc):
- don't send text tags if they are empty
- fix mem leak on error path
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (img_get_alpha_info):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- adds BGR32 and BGRA32 to ffmpegcolorspace
- (still bad colors, fixing it on next commit)
- helps with dvdec outputing BGR32
+ gst/playback/.cvsignore: Ignore more.
+ Original commit message from CVS:
+ * gst/playback/.cvsignore:
+ Ignore more.
-2005-01-08 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2005-01-07 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+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.
- * 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
+2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/Makefile.am:
+ interfaces.h is not used anymore so remove it from the build
+ process.
- * gst/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.
+2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
-2005-01-07 Gergely Nagy <algernon@bonehunter.rulez.org>
+ gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Add a "blink" pattern. Turn on the pain. Apologies. It's useful
+ for testing vertical refresh synchronization.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.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.
+ 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.
-2005-01-07 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ Makefile.am: Use new common/win32.mak.
+ Original commit message from CVS:
+ * Makefile.am:
+ Use new common/win32.mak.
- * 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).
+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).
-2005-01-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Fix against template (#150576).
+ 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.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/games/gstpuzzle.c: (draw_puzzle):
- don't draw a puzzle if either width or height of tiles would be 0.
+ gst-libs/gst/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).
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * 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
+ 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.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/games/gstvideoimage.c: (copy_hline_YUY2):
- fix stupid typo that borked copying on YUY2
+ 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.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/games/gstpuzzle.c: (draw_puzzle):
- fix edges when image sizes aren't multiples of tile sizes
+ 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.
-2005-01-06 Benjamin Otte <otte@gnome.org>
+2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
- * 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
+ * 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
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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)
+ 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.
-2005-01-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
- * ext/faad/gstfaad.c: (gst_faad_chain):
- Only increment timestamp if it's valid. Fixes raw AAC streams.
+ 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.
-2005-01-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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/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.
-2005-01-06 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.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-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?).
-2005-01-06 Paul Jack <pjack@sfaf.org>
+2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.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/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
- Fix segfault (#161667).
+2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
-2005-01-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ ChangeLog: remove conflict markers
+ Original commit message from CVS:
+ ChangeLog: remove conflict markers
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Fix framerate reporting.
+2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-05 Stephane LOEUILLET <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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>
+2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
- * configure.ac:
- back to cvs
+ gst/audioconvert/Makefile.am: Also link to libm.
+ Original commit message from CVS:
+ * gst/audioconvert/Makefile.am:
+ Also link to libm.
-=== release 0.8.7 ===
+2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.7, "Hyperspace"
+2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2005-01-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/playback/gstplaybasebin.c:
- Fix for #162924 - free caps after use, not before
+ 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.
-2005-01-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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/playback/gstplaybasebin.c:
- * gst/wavparse/gstwavparse.c:
- Fix for #154773 - fixes playback of small .wav files
+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.
-2005-01-03 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.
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+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).
- * gst/audioscale/gstaudioscale.c:
- Fix for #162819 - make audioscale reusable
- Fixes playback of more than one file with playbin/totem
+2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-plugins-base.doap: Oops, fix the release name.
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Oops, fix the release name.
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- * gst/ffmpegcolorspace/imgconvert.c:
- clean up the mess that made me cry and avoid needless duplication
+2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-29 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-plugins-base.doap: Add 0.10.15 release
+ Original commit message from CVS:
+ * gst-plugins-base.doap:
+ Add 0.10.15 release
- * gst/ffmpegcolorspace/imgconvert.c:
- give some indication of why we're segfaulting
+2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * 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>
+=== release 0.10.15 ===
- * ext/ogg/gstogg.c:
- * ext/ogg/gstogmparse.c:
- fix ogm[audio/video]parse plugin registration
- (riff won't load if bytestream is already loaded)
+2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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"
- * gst/audioconvert/gstchannelmix.c:
- fix for GLIB < 2.4
+2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/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.
- * Makefile.am:
- * configure.ac:
- disable docs again until it actually passes make distcheck.
+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).
-2004-12-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+ win32/common/: More missing symbols to export (fixes #493986).
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstcdda.def:
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstpbutils.def:
+ * win32/common/libgstrtp.def:
+ * win32/common/libgstrtsp.def:
+ * win32/common/libgsttag.def:
+ * win32/common/libgstvideo.def:
+ More missing symbols to export (fixes #493986).
- * gst/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.
+2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/fft/gstfftf32.c:
+ * gst-libs/gst/fft/gstfftf32.h:
+ * gst-libs/gst/fft/gstfftf64.c:
+ * gst-libs/gst/fft/gstfftf64.h:
+ * gst-libs/gst/fft/gstffts16.c:
+ * gst-libs/gst/fft/gstffts16.h:
+ * gst-libs/gst/fft/gstffts32.c:
+ * gst-libs/gst/fft/gstffts32.h:
+ * tests/check/libs/fft.c: (GST_START_TEST):
+ Remove the magnitude and phase calculation functions as these have
+ very special use cases and can't even be used for the spectrum
+ element. Also adjust the docs to mention some properties of the used
+ FFT implemention, i.e. how the values are scaled. Fixes #492098.
+
+2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
+ (finish_source):
+ Avoid crash when there are external subtitles (fixes #491722).
-2004-12-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Makefile.am:
- really fix dist
+ ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
+ 'Could not open resource for writing' is not an acceptable
+ error message when we can't open the audio device (see #492334),
+ even less so when we're trying to open it to record something.
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
- * configure.ac:
- * ext/speex/gstspeexdec.h:
- * ext/speex/gstspeexenc.h:
- Fixes #158382. Make speex plugin compatible with both 1.0 and 1.1.
- Fix detection code in configure.ac
+ win32/common/libgstrtp.def: Add some more missing symbols (#492813).
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * win32/common/libgstrtp.def:
+ Add some more missing symbols (#492813).
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Save position, so that queries give proper return values. Don't
- know how this could ever have worked before...
+ tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
+ Original commit message from CVS:
+ Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
+ * tests/check/elements/audioconvert.c: (verify_convert):
+ Add check to make sure that the out caps have a channel layout
+ set on them where they should have one.
-2004-12-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
- * configure.ac:
- Put additional LAME check inside the conditional. Fixes #152339
+ gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
+ Original commit message from CVS:
+ Patch by: Vincent Torri <vtorri at univ-evry dot fr>
+ * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
+ * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
+ Include our own _stdint.h instead of sys/types.h, makes MingW happy
+ (#492306).
+ * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
+ Use _pipe directly, GLib doesn't have a pipe() macro any longer
+ (it disappeared in GLib 2.14.0) (#492306).
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/sdp/gstsdpmessage.c:
+ Fix includes and LIBS for win32/Mingw (#492306).
+ * tests/examples/dynamic/addstream.c (pause_play_stream):
+ Use more portable g_usleep() instead of sleep() (#492306).
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan):
- Add some more debug. Fix logic error when setting movi offset
- while reading index.
+ gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
+ Original commit message from CVS:
+ Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
+ (gst_ring_buffer_parse_caps):
+ Return NULL instead of an enum that happens to be 0, fixes warning
+ on MSVC (#492114).
+ * gst-libs/gst/audio/gstringbuffer.h:
+ No trailing commas in enum list (for gcc-2.9x).
+ * gst/videotestsrc/videotestsrc.c: (random_char):
+ Make information loss explicit instead of implicitly truncating to
+ eight bits via the return value. Fixes runtime error on MSVC when
+ using the debug CRT (#492114).
+ * win32/common/config.h.in:
+ Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
+ * win32/common/libgstinterfaces.def:
+ * win32/common/libgstrtp.def:
+ Export a few more symbols (#492114).
-2004-12-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan), (gst_avi_demux_handle_seek),
- (gst_avi_demux_process_next_entry):
- Add some debugging. Better detection of broken indexes and the
- accompanying index recovery. No infinite loops on state changes
- when we're still in our loopfunction.
+ gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/audio.h:
+ Readd the deprecation guards, but preserve compilability.
-2004-12-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Fix up.
+ gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
+ (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
+ Preserve channel layout when fixating the number of channels in the
+ output caps, or make sure there's a suitable channel position layout
+ set on the caps if required. Fixes #430677.
-2004-12-22 Archana Shah <archana.shah@wipro.com>
+2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_volume):
- Normalizing the value before setting
- (gst_sunaudiomixer_get_volume):
- Normalizing the value after getting. Fixes bug# 161980
+2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-22 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ 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.
- * Makefile.am: Make sure docs gets disted
- * docs/Makefile.am: Make sure all needed files get disted
- * gst-plugins.spec.in: latest updates
+2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-22 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- Revert patch 1.38 as clock distribution over schedulers does
- not work correcly in the core yet.
+2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * sys/oss/README: remove this file, which predates my birth
- (and which content is by far outdated)
+2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-20 Stefan Kost <ensonic@users.sf.net>
+ 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.
- * Makefile.am:
- * configure.ac:
- * docs/Makefile.am:
- * docs/libs/Makefile.am:
- * docs/libs/gst-plugins-libs-docs.sgml:
- * docs/libs/gst-plugins-libs-sections.txt:
- * docs/libs/tmpl/gstgconf.sgml:
- * docs/upload.mak:
- * docs/version.entities.in:
- Added boilerplate gtk-doc files for plugin-libs documentation.
+2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ tests/icles/stress-playbin.c: Fix the build.
+ Original commit message from CVS:
+ * tests/icles/stress-playbin.c:
+ Fix the build.
- * gst/auparse/gstauparse.c: fix int and float audio caps
+2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- * sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
- g_assert() can be a macro, don't use #ifdef inside it.
+2007-10-30 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>
-2004-12-19 Edward Hervey <bilboed@bilboed.com>
+ * 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/videorate/gstvideorate.c: (gst_videorate_blank_data),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_change_state):
- Event handling (fixes #159986).
+ * gst-plugins-base.spec.in:
+ update spec file
+ Original commit message from CVS:
+ update spec file
+
+2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
+ (gst_decode_bin_set_subs_encoding),
+ (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
+ (gst_decode_bin_get_property), (analyze_new_pad):
+ Move subtitle encoding property to decodebin2 so that it can set the
+ property value on all elements that it autoplugs and that require it.
+ Make caps refcounting more consistent in get/set.
+ * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
+ (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
+ (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
+ (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
+ (proxy_autoplug_continue_signal),
+ (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
+ (make_decoder):
+ Proxy properties and relevant signals from the internal decodebin.
+ Make properties MT safe.
+
+2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
+ * gst-libs/gst/tag/tags.c:
+ Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
+ GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
+ * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
+ Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
+ * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
+ (gst_tag_to_vorbis_comments):
+ Map new SORTNAME tags (these tags aren't even semi-official, so I'm
+ just mapping everything I found in the wild) (#414539).
-2004-12-19 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
+ Original commit message from CVS:
+ Inspired by patch of: René Stadler <mail at renestadler dot de>
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
+ (gst_decode_bin_autoplug_continue),
+ (gst_decode_bin_autoplug_factories),
+ (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
+ (find_compatibles):
+ * gst/playback/gstplay-marshal.list:
+ Remove the autoplug-sort signal and replace it with a binding friendly
+ autoplug-select signal.
+ Add an autoplug-factories signal that can be used to generate a list of
+ factories to try to autoplug.
+ Add the GstPad to the autoplugging signal args as it might be needed to
+ make a good factory selection.
+ Fix up the marshallers for this. Fixes #407282.
+
+2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't abort with an assertion if we receive a seek event with
+ a start type of NONE (see launchpad bug #155878).
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add BLZ0 (Blizzard's version of DivX) fourcc.
+2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
+ (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
+ (gst_ximagesink_change_state), (gst_ximagesink_reset):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
+ (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
+ (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
+ Make sure that before we clean up the X resources, we shutdown and join
+ the event thread.
+ Also make sure the event thread does not shut down immediatly after
+ startup because the running variable is not yet correctly set.
+ Fixes #378770.
+
+2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (new_pad), (type_found):
+ Make the window for a race in typefind and shutting down smaller until
+ we figure out the right locking here. Avoids #485753 usually.
+ * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
+ Remove unneeded lock causing a race in typefind and shutting down.
+ Fixes #485753.
+ * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
+ Also remove sinks when going to NULL because we might not complete the
+ state change to PAUSED, causing the PAUSED->READY state change not to
+ happen.
+
+2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
+ Also explicitly release the ringbuffer when going to NULL because it
+ is required in the setcaps function, before the state change to PAUSED
+ completes.
+
+2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/icles/: Does what it says on the tin.
+ Original commit message from CVS:
+ * tests/icles/.cvsignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/stress-playbin.c:
+ Does what it says on the tin.
-2004-12-18 David Schleef <ds@schleef.org>
+2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tta/gstttadec.c: (gst_tta_dec_link): And yet another
- portability fix.
+ 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.
-2004-12-18 David Schleef <ds@schleef.org>
+2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tta/ttadec.h: Disable some header code that isn't used
- and clearly isn't portable.
+ 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.
-2004-12-18 David Schleef <ds@schleef.org>
+2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/ffmpegcolorspace/imgconvert.c: (get_pix_fmt_info),
- (avcodec_get_chroma_sub_sample), (avcodec_get_pix_fmt_name),
- (avcodec_get_pix_fmt), (avpicture_layout),
- (avcodec_get_pix_fmt_loss), (avg_bits_per_pixel), (img_copy),
- (get_convert_table_entry), (img_convert), (img_get_alpha_info):
- Fix code to not use GCC extensions (and c99 extensions that
- Forte does not like.)
+ * ChangeLog:
+ Remove stray character from the changelog.
+ Original commit message from CVS:
+ Remove stray character from the changelog.
-2004-12-19 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_link),
- (gst_deinterlace_chain):
- Rowstride fixes. Fixes #161039.
- * gst/videocrop/gstvideocrop.c: (gst_video_crop_init),
- (gst_video_crop_get_property), (gst_video_crop_add_to_struct_val),
- (gst_video_crop_getcaps), (gst_video_crop_link),
- (gst_video_crop_i420), (gst_video_crop_chain),
- (gst_video_crop_change_state):
- Rework of negotiation. Actually works now. Fixes #158650.
+2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_sint):
- That was very stupid.
+2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix possible crasher.
+2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't error out when a buggy downstream element doesn't
+ handle the newsegment event we send properly (especially
+ not without posting a meaningful error message on the
+ bus). See bug #471370 and launchpad bug #136264.
- * gst/matroska/matroska-demux.c: (gst_matroska_ebmlnum_uint),
- (gst_matroska_ebmlnum_sint), (gst_matroska_demux_parse_blockgroup):
- Lace sizes can be zero.
+2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain):
+ Use new basesink method to make our EOS drain interruptable.
+
+2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ Fix silly search-replace oversight.
+
+2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
+
+ gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
+ Original commit message from CVS:
+ Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+ * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
+ (gst_basertppayload_set_outcaps):
+ Fix caps memleak. Fixes #484989.
+
+2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
+ Fix debug output.
+
+2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Also handle the case where there is no clock set on the audio source,
+ like in the unit tests.
+
+2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtppayloads.c:
+ Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
+ to avoid compiler warnings
+
+2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (type_found),
+ (gst_decode_bin_change_state):
+ * gst/playback/gstdecodebin2.c: (type_found),
+ (gst_decode_bin_change_state):
+ Don't disconnect the have_type signal because we never reconnect it
+ later on. Instead keep a variable to see if we already detected a type.
+
+2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
+ * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
+ (type_found):
+ Unlink the signal handler when we found the type, we're not going to do
+ anything sensible with more type_found signals anyway.
+
+2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
+ Original commit message from CVS:
+ * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
+ Use GIO function to get a list of supported URI schemes instead of
+ hard coding something.
+
+2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gsttagdemux.c:
+ Don't leak caps.
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/musepack/gstmusepackdec.cpp:
- Fetch error return values. Fixes #161624.
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_data):
- Really EOS.
+ 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.
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
- Work for truncated (unfinished download etc.) files. Fixes #160514.
+ 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.
-2004-12-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- Fix for integer overflow. Makes #156001 not crash. Probably masks
- the real bug.
+ gst-libs/gst/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.
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/ac3parse/gstac3parse.c: (plugin_init):
- Parsers never have ranks. Fixes #159651.
+ tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
+ Original commit message from CVS:
+ * tests/check/pipelines/simple-launch-lines.c:
+ Print message name and not just number.
-2004-12-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_async_play):
+ When slaved to the clock, don't try to align a sample with the previous
+ one when going to PLAYING again.
-2004-12-16 David Schleef <ds@schleef.org>
+2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac: look for musepack headers as musepack/*.h
- (fixes #159847)
- * ext/musepack/gstmusepackdec.h: use <musepack/*.h>
- * ext/musepack/gstmusepackreader.h: same
+ tests/examples/snapshot/snapshot.c: Fix the build.
+ Original commit message from CVS:
+ * tests/examples/snapshot/snapshot.c:
+ Fix the build.
-2004-12-17 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_auds_with_data):
- Read extradata correctly (fixes #155879).
+2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-16 David Schleef <ds@schleef.org>
+ gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
+ Original commit message from CVS:
+ * gst-libs/gst/floatcast/floatcast.h:
+ Don't include config.h in an installed public header, this
+ might break compilation of applications that don't have such
+ a header and doesn't necessarily do what it's supposed to do
+ anyway (ie. check for the lrint/lrintf defines) (#442065).
+ Add docs for the various macros and document how this header
+ has to be used (link against libm, etc.); add a few FIXMEs;
+ include math.h for non-c99 code path. Based on patch by
+ Jan Schmidt.
- * gst/audioscale/gstaudioscale.c: allow passthru of >2 channel
- audio. does _not_ attempt or allow conversion unless channels
- is 1 or 2.
+2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-12-16 Christian Fredrik Kalager Schaller <uraeus@gnome.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.
- * tools/gst-launch-ext-m.m: fix mpeg and vob pipelines
+2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 David Schleef <ds@schleef.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
- * gst/audioscale/gstaudioscale.c: the resample library only
- handles 1 or 2 channels. Change caps to compensate.
+2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * po/LINGUAS:
+ add languages
+ Original commit message from CVS:
+ add languages
- * 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...
+2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/pl.po: Added Polish translation.
+ Original commit message from CVS:
+ translated by: Jakub Bogusz <qboosh@pld-linux.org>
+ * po/pl.po:
+ Added Polish translation.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add h264.
+2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/fi.po: Added Finnish translation.
+ Original commit message from CVS:
+ translated by: Ilkka Tuohela <hile@iki.fi>
+ * po/fi.po:
+ Added Finnish translation.
- * gst-libs/gst/audio/Makefile.am:
- Try to fix buildbot.
+2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * gst/tcp/gstmultifdsink.c:
- Clean up and uniformize debugging.
+2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-16 Edward Hervey <bilboed@bilboed.com>
+ po/da.po: Added Danish translation.
+ Original commit message from CVS:
+ translated by: Mogens Jaeger <mogens@jaeger.tf>
+ * po/da.po:
+ Added Danish translation.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ po/zh_CN.po: Added Chinese (simplified) translation.
+ Original commit message from CVS:
+ translated by: Funda Wang <fundawang@linux.net.cn>
+ * po/zh_CN.po:
+ Added Chinese (simplified) translation.
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/bg.po: Added Bulgarian translation.
+ Original commit message from CVS:
+ translated by: Alexander Shopov <ash@contact.bg>
+ * po/bg.po:
+ Added Bulgarian translation.
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_pad_link):
- Fix memleak (#154815).
+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.
-2004-12-16 James Bowes <bowes@cs.dal.ca>
+2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ Ignore registries in any format.
+ Original commit message from CVS:
+ * docs/plugins/.cvsignore:
+ * tests/check/.cvsignore:
+ Ignore registries in any format.
- * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init),
- (gst_musicbrainz_init), (gst_musicbrainz_chain),
- (gst_musicbrainz_set_property), (gst_musicbrainz_get_property):
- * ext/musicbrainz/gsttrm.h:
- Add support for using a proxy server when getting a trm id from
- the MusicBrainz database (#149613).
+2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-16 Christophe Fergeau <teuf@gnome.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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstdecodebin.c: (new_pad), (close_link):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Fix memleaks (#157233).
+ 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.
-2004-12-16 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Improve debug logs a bit and be more verbose if things go wrong.
- * gst-libs/gst/resample/resample.c: (gst_resample_close):
- * gst-libs/gst/resample/resample.h:
- * gst/audioscale/gstaudioscale.c:
- Fix memleak (#159215).
+2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-16 Toni Willberg <toniw@iki.fi>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * sys/oss/gstosselement.c: (gst_osselement_probe_caps):
- * sys/oss/oss_probe.c: (main):
- Check for mono/stereo support (similar to samplerate probing),
- fixes #159433. Also add missing copyright header to oss_probe.c.
+2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-15 David Schleef <ds@schleef.org>
+ 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.
- * configure.ac: add audioresample and cairo plugins. Remove
- HAVE_MMX stuff, because it's not used.
- * ext/Makefile.am: same
- * ext/audioresample/Makefile.am: You are not ready for an
- audio resampling element based on audioresample.
- * ext/audioresample/gstaudioresample.c:
- * ext/audioresample/gstaudioresample.h:
- * ext/cairo/Makefile.am: You are not ready for overlay elements
- based on cairo. Don't look too closely, these elements kinda
- suck right now.
- * ext/cairo/gstcairo.c: new
- * ext/cairo/gsttextoverlay.c: new
- * ext/cairo/gsttextoverlay.h: new
- * ext/cairo/gsttimeoverlay.c: new
- * ext/cairo/gsttimeoverlay.h: new
- * gst-libs/gst/media-info/media-info-priv.h: fix compile
- problem with compilers that don't support variadic macros.
+2007-09-12 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).
-2004-12-15 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+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.
- Reviewed by: David Schleef <ds@schleef.org>
+2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
- * 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):
+ sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler dot de>
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximage_buffer_class_init):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_class_init):
+ Correctly chain up finalize with the parent class to prevent
+ memory leaks. Fixes #474880.
-2004-12-14 David Schleef <ds@schleef.org>
+2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * gst/festival/gstfestival.c: (gst_festival_chain): Set the
- output rate to 16000. Should fix #160235.
+ Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
+ Original commit message from CVS:
+ * gst/volume/gstvolume.c: (volume_choose_func):
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ Revert the latest change: floating point samples are allowed to
+ have any value, not only values in the range [-1,1]. Thanks to Andy
+ Wingo for noticing.
+ Also fix processing of int32 samples with volumes > 4 by making the
+ unity value smaller which prevents overflows.
-2004-12-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Add typefinding for mpeg2 pes streams
+ Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * tests/check/libs/rtp.c:
+ Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
-2004-12-13 David Schleef <ds@schleef.org>
+2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
- * 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.
+ gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
+ Original commit message from CVS:
+ Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ Fix up GstRTPHeader helper struct so that compilers will not under
+ any circumstances add padding in between our fields, as currently
+ happens with MSVC on win32, because that would lead to us sending
+ out RTP payloads with broken RTP headers (#471194).
+ Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/rtp.c:
+ Add some simple unit tests for GstRTPBuffer. Some are disabled
+ because the code tested still needs fixing (set_csrc() does not work).
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
- * gst-libs/gst/riff/riff-media.c:
- forgot to add h2.64 to avidemux template caps
+ * gst-plugins-base.spec.in:
+ update spec file to include latest RTSP libraries and headers and more
+ Original commit message from CVS:
+ update spec file to include latest RTSP libraries and headers and more
+
+2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ win32/: Add rtsp enumtypes (#474384) and update others.
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ * win32/common/gstrtsp-enumtypes.c:
+ * win32/common/gstrtsp-enumtypes.h:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ Add rtsp enumtypes (#474384) and update others.
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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 :(
+ configure.ac: Fix configure check for HAVE_LIBXML_HTML.
+ Original commit message from CVS:
+ * configure.ac:
+ Fix configure check for HAVE_LIBXML_HTML.
-2004-12-13 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
+ Original commit message from CVS:
+ * tests/check/libs/.cvsignore:
+ Ignore more, in case the build bots work again one day.
-2004-12-12 Christian Fredrik Kalager Schaller <christian at fluendo dot com>
+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.
- * configure.ac: Update polyaudio requirement to 0.7
- * ext/polyp/polypsink.c: (create_stream): add patch from iain (158258)
+2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/playback/gststreaminfo.c: Fix build.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c:
+ Fix build.
- * gst/interleave/deinterleave.c:
- fix my name's spelling! :)
+2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * AUTHORS ChangeLog
- * gst/auparse/gstauparse.c
- * gst/interleave/deinterleave.c
- * gst/law/:
- alaw-decode.c alaw-encode.c
- mulaw-decode.c mulaw-encode.c
- * gst/oneton/gstoneton.c
- * sys/osxaudio/:
- gstosxaudioelement.c gstosxaudiosink.c gstosxaudiosrc.c
- * sys/osxvideo/:
- cocoawindow.h cocoawindow.m
- osxvideosink.h osxvideosink.m
+2007-09-04 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.
- put the same mail address for Zaheer Abbas Merali everywhere
+2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
-2004-12-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (get_relative):
- Check for non-NULL before accessing member (end-of-chain).
+2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * configure.ac: move GCONF macro outside conditional for the am
- conditional. Fixes #160439
+2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-08 David Schleef <ds@schleef.org>
+ 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.
- * tools/gst-visualise-m.m: Switch to elements that currently
- exist.
+2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
- We love wrong commas.
+2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_src_query):
- Don't set DEFAULT, unsupported - makes length display incorrectly
- in some cases.
+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.
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+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/monoscope/README: remove blurb about files being GPL
- * gst/monoscope/gstmonoscope.c: Change license field to LGPL
- * gst/monoscope/monoscope.c: Change license to BSD with explanation
- monoscope is now effectively LGPL licensed
+2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-07 Christian Fredrik Kalager Schaller <uraeus@gnome.org>
+ * 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/monoscope/README: Update information to be more correct
- * gst/monoscope/convolve.c: Relicense to LGPL
- * gst/monoscope/convolve.h: Relicense to LGPL
+2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-12-06 Arwed v. Merkatz <v.merkatz@gmx.net>
+ gst-libs/gst/audio/audio.c: Clarify the docs a little.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Clarify the docs a little.
- * 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
+2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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.
- * 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)
+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.
-2004-12-06 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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.
+2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/apetag/apedemux.c: (gst_ape_demux_parse_tags),
- (gst_ape_demux_stream_init):
- Forward tags, too.
+2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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)
- * gst/apetag/apedemux.c: (gst_ape_demux_stream_init):
- Let's make sure we're done typefinding when detecting tags.
+2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiorate/gstaudiorate.c: Debug output fixes.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Debug output fixes.
+ * tests/check/elements/audiorate.c: (do_perfect_stream_test),
+ (GST_START_TEST):
+ Change the number of buffers used; 500 is too many and leads to
+ timeouts.
- * ext/musepack/gstmusepackdec.cpp:
- There's also floating point libmusepacks.
+2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-12-03 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Printf format fixes (#465028).
+ Original commit message from CVS:
+ * gst/playback/gstqueue2.c:
+ * gst/videorate/gstvideorate.c:
+ Printf format fixes (#465028).
- * ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
- (gst_faad_chanpos_to_gst), (gst_faad_chain):
- Set DURATION even if source buffer didn't. Also use increasing
- timestamps.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Block_align can have larger values than 8192.
+2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
-2004-12-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ If we have a large (> 1 second) discontinuity, push a series of
+ smaller buffers rather than a single very large buffer. Avoids
+ unreasonably large single buffer allocations when encountering a
+ large gap.
+ * tests/check/elements/audiorate.c: (GST_START_TEST),
+ (audiorate_suite):
+ Add a test for this.
- * 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.
+2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Fixes: #465015
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_remove_probe), (queue_threshold_reached):
+ Patch by: Josep Torra Valles <josep@fluendo.com>
+ Fixes: #465015
+ Make sure we remove the check_queues buffer probe from the
+ correct queue to avoid racily going back to "buffering 99%" when
+ buffering is actually complete.
+ Also, fix the spelling of Josep's surname in the ChangeLog.
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix logic bug.
+2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/ogg/gstoggmux.c: Do not leak oggmux instance.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ Do not leak oggmux instance.
+ * ext/vorbis/vorbisenc.c:
+ Also log values.
- * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
- Yay, another one.
+2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/: Updated translations.
+ Original commit message from CVS:
+ * po/hu.po:
+ * po/it.po:
+ * po/nl.po:
+ * po/uk.po:
+ * po/vi.po:
+ Updated translations.
- * 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).
+2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
-2004-12-01 Christophe Fergeau <teuf@gnome.org>
+ ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
+ Original commit message from CVS:
+ patch by: Yang Hong <hongyang@redflag-linux.com>
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ Add 'silent' property to GstTimeOverlay. Fixes #462979
- * ext/mad/gstmad.c: (mpg123_parse_xing_header): fix xing header
- detection on mono and stereo mp3 files.
+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
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+=== release 0.10.14 ===
- * 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).
+2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ Release 0.10.14
+ Original commit message from CVS:
+ Release 0.10.14
- * ext/faad/gstfaad.c: (gst_faad_srcconnect), (gst_faad_chain):
- Oops, remove debug.
+2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
+ * 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
- * gst/law/alaw-decode.c: (alawdec_getcaps):
- * gst/law/mulaw-decode.c: (mulawdec_getcaps):
- Prevent warnings when negotiating caps (fixes #159338).
+ 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.
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_chain):
- Remove old leftover that shouldn't be there...
+ 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.
-2004-12-01 Sebastien Cote <sc5@hermes.usherb.ca>
+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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
- Don't forward DISCONT events (fixes #159684).
+ tests/: We should use $(LIBM).
+ Original commit message from CVS:
+ * tests/examples/volume/Makefile.am:
+ * tests/icles/Makefile.am:
+ We should use $(LIBM).
-2004-12-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.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.
+ tests/icles/Makefile.am: This needs -lm.
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ This needs -lm.
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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.
+2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ Fix 'make check' build against core CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ Fix 'make check' build against core CVS.
- * 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.
+2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/: Make gtk-doc happy.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Make gtk-doc happy.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Actually leave the loop if we failed to sync. Don't crash.
+2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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. :-(.
+2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-28 Benjamin Otte <otte@gnome.org>
+ 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/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
- more overwriting protection due to modifying channels one by one
- instead of all at once
+2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
+ (img_get_alpha_info):
+ Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
+ of the existing BGRA32 and RGBA32 formats with the alpha at the other
+ end of the word. Partially fixes #451908
+
+2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/: Simplify --extra-dir as gtkdoc scans recursively.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/plugins/Makefile.am:
+ Simplify --extra-dir as gtkdoc scans recursively.
-2004-11-28 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/audioconvert/gstchannelmix.c:
- (gst_audio_convert_fill_normalize):
- Normalize using absolute values.
+2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-28 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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>
+2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ gst/playback/gstplaybasebin.c: Small debug improvement.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
+ Small debug improvement.
+ * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
+ (plugin_init):
+ Tweak the rate estimation period.
+ When calculating the buffer filledness in rate estimation mode, don't
+ mix it with other metrics.
+
+2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
+ (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
+ When creating the groups, allow for a 5 second, unlimited buffers
+ preroll phase after which we expose the group.
+ When the group is exposed, use a small number of buffers up to a 2
+ second limit. Also disconnect the overrun signal from multiqueue when we
+ exposed the group because it is not needed anymore.
+
+2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
+ Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
+ to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
+ (#451707); also, output some debugging info when dealing with
+ freeform strings.
+ * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
+ Add unit test for the above.
-2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
+2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2004-11-27 Christophe Fergeau <teuf@gnome.org>
+2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
+ Don't error out on 0-sized packets, just emit a warning because this is
+ not a fatal error. Fixes #451145.
+
+2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/plugins/: Update docs with caps info.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ Update docs with caps info.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
- Don't forget bass if it's there. Else left channel is silent...
+2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ po/POTFILES.in: Add more files with translatable strings (#450875).
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Add more files with translatable strings (#450875).
- * 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.
+2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/a52dec/gsta52dec.c: (gst_a52dec_channels),
- (gst_a52dec_change_state):
- Advertise that we can do surround sound.
+2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-11-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ MAINTAINERS: Updating all the maintainers files
+ Original commit message from CVS:
+ * MAINTAINERS:
+ Updating all the maintainers files
- * 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!).
+2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-27 Christophe Fergeau <teuf@gnome.org>
+ 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/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
- call parent dispose method
+2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
-2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
+ 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/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.
+2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
-2004-11-27 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * ChangeLog:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * win32/common/config.h:
+ gst-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ 2007-06-19 Andy Wingo <wingo@pobox.com>
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_init): Enable pull-mode operation.
- * 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.
+2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Fix ilog mask range overflow.
+2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ po/vi.po: Update translations.
+ Original commit message from CVS:
+ * po/vi.po:
+ Update translations.
-2004-11-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
- * 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.
+ 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.
-2004-11-26 Christophe Fergeau <teuf@gnome.org>
+2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_audio_element): Removed 2 obsolete comments
+ gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
+ Original commit message from CVS:
+ * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
+ Update tmpbuf for all neccesary rows, not just one, as is required
+ when downscaling.
+ Fixes #402076.
-2004-11-26 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
- * 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)
+ tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
+ (eos_buffer_probe):
+ Add a test that ensures we set DELTA_UNIT on all non-header,
+ non-video buffers, if we have a video stream.
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
+ (gst_ogg_mux_process_best_pad):
+ Move setting delta_pad to earlier, where we inspect all pads, so
+ that leading audio pages don't get DELTA_UNIT unset if they come
+ before the first DELTA_UNIT from video pages. Fixes the newly-added
+ test. Fixes #385527.
-2004-11-25 Christophe Fergeau <teuf@gnome.org>
+2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
- (gst_play_bin_set_property), (gen_video_element),
- (gen_audio_element):
- Refcounting fixes for provided audio-/videosinks.
+2007-06-13 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.
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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.
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
- (gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps):
- * ext/alsa/gstalsaplugin.c: (plugin_init):
- * ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
- (gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
- (gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
- (gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
- (gst_faad_change_state), (plugin_init):
- * ext/faad/gstfaad.h:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- * gst-libs/gst/audio/Makefile.am:
- * gst-libs/gst/audio/audio.c: (plugin_init):
- * gst-libs/gst/audio/multichannel.c:
- (gst_audio_check_channel_positions),
- (gst_audio_get_channel_positions),
- (gst_audio_set_channel_positions),
- (gst_audio_set_structure_channel_positions_list),
- (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
- (gst_audio_fixate_channel_positions):
- * gst-libs/gst/audio/multichannel.h:
- * gst-libs/gst/audio/testchannels.c: (main):
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_class_init), (gst_audio_convert_init),
- (gst_audio_convert_dispose), (gst_audio_convert_getcaps),
- (gst_audio_convert_parse_caps), (gst_audio_convert_link),
- (gst_audio_convert_fixate), (gst_audio_convert_channels):
- * gst/audioconvert/plugin.c: (plugin_init):
- Surround sound support.
+2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
- Fix position for discont if we're close as well. Nitpicking, but
- saves a few milliseconds of extra waiting or skipping.
+2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/playback/gstdecodebin.c: (gst_decode_bin_factory_filter):
- We sometimes need parsers for playback, so add those too.
+2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * configure.ac:
- * gst/apetag/Makefile.am:
- * gst/apetag/apedemux.c: (gst_ape_demux_get_type),
- (gst_ape_demux_base_init), (gst_ape_demux_class_init),
- (gst_ape_demux_init), (gst_ape_demux_get_src_formats),
- (gst_ape_demux_get_src_query_types),
- (gst_ape_demux_handle_src_query), (gst_ape_demux_get_event_mask),
- (gst_ape_demux_handle_src_event), (gst_ape_demux_handle_event),
- (gst_ape_demux_typefind_peek), (gst_ape_demux_typefind_get_length),
- (gst_ape_demux_typefind_suggest), (gst_ape_demux_typefind),
- (gst_ape_demux_parse_tags), (gst_ape_demux_stream_init),
- (gst_ape_demux_stream_data), (gst_ape_demux_loop),
- (gst_ape_demux_change_state):
- * gst/apetag/apedemux.h:
- * gst/apetag/apetag.c: (plugin_init):
- * gst/typefind/gsttypefindfunctions.c: (apetag_type_find),
- (plugin_init):
- APE v1/2 tag reader plus typefind function.
-
-2004-11-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/: 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.
- * configure.ac:
- back to CVS
+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
-=== release 0.8.6 ===
+=== release 0.10.13 ===
-2004-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-plugins-base.doap:
+ * win32/common/config.h:
+ * 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?"
+
+2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
* po/af.po:
* po/az.po:
* po/cs.po:
+ * po/de.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
- releasing 0.8.6, "IOU Love"
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c:
- Fix unplayable files error handling. Fixes #158365
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c:
- Fix broken mp3 typefinding. Fixes #158375
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/ogg/gstoggdemux.c:
- Fix sync on broken files. Fixes #158976
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Edward Hervey <bilboed@bilboed.com>
-
- * ext/libpng/gstpngenc.c:
- Copy over buffer properties. Fixes #158832
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Tim-Philipp Müller <t.i.m@zen.co.uk>
-
- * ext/dvdread/dvdreadsrc.c:
- Fixes invalid reads (#158462)
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- Probe less and cache it. Fixes #159187.
-
-2004-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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>
-
- * configure.ac:
- Fix mpeg2enc configure check (similar to mplex check below).
-
-2004-11-14 Koop Mast <kwm@rainbow-runner.nl>
-
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- Fix for gcc-2.95 (fixes #158221).
-
-2004-11-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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.
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- Re-add clock distribution hack (until new core is released).
- Fixes #158125.
+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.
-2004-11-13 Arwed v. Merkatz <v.merkatz@gmx.net>
- * configure.ac:
- fix mplex configure check segfaulting on some systems (bug #140994)
+2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
-2004-11-13 Benjamin Otte <otte@gnome.org>
+ ext/gnomevfs/gstgnomevfssink.*: see #412648.
+ Original commit message from CVS:
+ Patch by: Mark Nauwelaerts <manauw at skynet be>
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
+ (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
+ (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Fix position reporting, especially after a seek (from upstream),
+ see #412648.
- * 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
+2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-13 Jan Schmidt <thaytan@mad.scientist.com>
+ ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ Repair umlaut.
- * 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>
+2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/volume/gstvolume.c:
- Allow buffer-frames=0.
+ 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.
-2004-11-12 Iain <iaingnome@gmail.com>
+2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
- * configure.ac: Check for polypaudio
+ 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.
- * ext/Makefile.am: Build the polyp dir
+2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
- * ext/polyp: The polypsink sources.
+ ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_data_packet):
+ Don't look inside 0-length packets (which indicate duplicated
+ frames)
-2004-10-30 Iain <iaingnome@gmail.com>
+2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Small cleanups.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_read_sector):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_create):
+ Small cleanups.
+ * ext/theora/theoradec.c: (theora_dec_sink_event):
+ Fix typo.
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Add some FIXME
+ * gst/playback/gstdecodebin.c: (queue_underrun_cb):
+ And some debug info when a FIXME path is hit.
+
+2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_class_init),
+ (gst_base_rtp_audio_payload_init),
+ (gst_base_rtp_audio_payload_finalize),
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer),
+ (gst_base_rtp_payload_audio_handle_event):
+ Some cleanups, remove minptime property as it is now in the parent
+ class.
+ Override parent class event function.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ (gst_basertppayload_class_init), (gst_basertppayload_init),
+ (gst_basertppayload_event), (gst_basertppayload_set_property),
+ (gst_basertppayload_get_property):
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ Add min-ptime property.
+ Add handle-event vmethod. Fixes #415001.
+
+2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/interleave/interleave.c (interleave_unlink): Change the src pads
- caps to reflect the new number of channels.
+ * gst-plugins-base.spec.in:
+ update spec
+ Original commit message from CVS:
+ update spec
-2004-11-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.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-libs/gst/audio/gstbaseaudiosink.c
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c
+ (gst_base_audio_sink_change_state):
+ Fix typo in comment.
+ * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
+ free_dynamics, pad_probe, close_pad_link, try_to_link_1,
+ get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
+ close_link):
+ * gst/playback/gstplaybin.c (gst_play_bin_set_property,
+ gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
+ Remove trailing whitespaces in comments.
+ * gst/volume/Makefile.am:
+ Fix tabs.
-2004-11-11 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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).
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
- No warnings (#157986).
+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.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- Prefer apev1/2 and id3v1 (at end of file) over musepack.
+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.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream):
- Signal no-more-pads (so it works in playbin).
+ 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.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/musepack/gstmusepackreader.cpp:
- Workaround for older core.
+ 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.
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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).
+ 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.
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+ 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.
-2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
- Fix compilation if HAVE_XVIDEO is not defined
+ * 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.
-2004-11-11 Jan Schmidt <thaytan@mad.scientist.com>
+2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
- * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
- (gst_goom_dispose), (gst_goom_sinkconnect), (gst_goom_chain),
- (gst_goom_change_state), (plugin_init):
- Use the bytestream adapter so goom doesn't depend on the input
- buffer size.
- Add a debug category
+ 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
-2004-11-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.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).
+ 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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
- * 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).
+ 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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Don't assert (#157853).
+ 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.
-2004-11-10 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event),
- Fix bytes/samples confustion.
- (gst_alsa_sink_mmap), (gst_alsa_sink_loop):
- Fix for underrun (#144389).
+ ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_collect_chain_info):
+ If there is a stream in a chain without any data packets, ignore the
+ stream in the total length calculations. Might be related to #436820.
-2004-11-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- Disable halfway-seek for pending release (since it needs a new
- core release).
+ 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.
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
- * sys/v4l/gstv4lsrc.c:
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c:
- add autoprobe-fps property so we can separate autoprobing parts
+ 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!
-2004-11-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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.
-2004-11-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- Submitted by: Luca Ferretti <elle.uca@infinito.it>
+ * 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.
- * po/LINGUAS:
- * po/it.po:
- Add Italian
+2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_reset),
+ (theora_dec_sink_event), (theora_handle_comment_packet),
+ (theora_handle_type_packet), (theora_dec_change_state):
+ Don't push events (newsegment, tags) before initialising the
+ decoder.
+ This is neccesary for seeking to work correctly in gnonlin.
- * gst/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..
+2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-08 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/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
- We can remove this hack now.
+2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
+ Minimal check for volume's GstController usability; also another
+ test for #422295.
- * gst/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.
+2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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: (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.
+2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_add_track):
+ Fix build when disabling asserts.
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_set_property), (gst_alpha_sink_link),
- (gst_alpha_set_ayuv), (gst_alpha_set_i420),
- (gst_alpha_chroma_key_ayuv), (gst_alpha_chroma_key_i420),
- (gst_alpha_init_params), (gst_alpha_chain):
- Implement alpha functions for AYUV too, this increases
- accuracy quite a bit.
+2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@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).
- * gst/ffmpegcolorspace/avcodec.h:
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (shrink12), (img_get_alpha_info), (deinterlace_line),
- (deinterlace_line_inplace):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Added AYUV colorspace and handle RGBA a bit more respectful.
+2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
-2004-11-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Actually always send a discont (cornercase when resending the
- same serial-tagged chain twice).
+2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-08 Julien MOUTTE <julien@moutte.net>
+ 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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
- (gst_ximagesink_finalize):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_finalize): Some more cleanups, leaks fixed and checks.
+2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
- Don't segfault on NULL data.
+2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-08 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/playback/gstdecodebin.c: (unlinked):
- * gst/playback/gstplay-marshal.list:
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
- (gst_play_base_bin_init), (group_create), (get_active_group),
- (get_building_group), (group_destroy), (group_commit),
- (queue_overrun), (remove_groups), (add_stream), (unknown_type),
- (add_element_stream), (no_more_pads), (probe_triggered),
- (preroll_unlinked), (new_decoded_pad), (removed_decoded_pad),
- (state_change), (setup_source), (gst_play_base_bin_get_property),
- (gst_play_base_bin_change_state), (gst_play_base_bin_add_element),
- (gst_play_base_bin_link_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
- (remove_sinks), (setup_sinks), (gst_play_bin_change_state):
- Add support for chained ogg files. Prepare for playlist
- support. This patch introduces the concept of pad groups, which
- together compose one playable media file.
+2007-04-29 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).
-2004-11-07 David Schleef <ds@schleef.org>
+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).
- * testsuite/gst-lint: Check for pad templates that aren't statically
- scoped.
+2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * configure.ac:
- * ext/Makefile.am:
- * ext/musepack/Makefile.am:
- * ext/musepack/gstmusepackdec.cpp:
- * ext/musepack/gstmusepackdec.h:
- * ext/musepack/gstmusepackreader.cpp:
- * ext/musepack/gstmusepackreader.h:
- Add musepack decoder.
- * ext/faad/gstfaad.c: (gst_faad_base_init):
- Make pad templates static.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (plugin_init):
- Add musepack typefinder, make mp3 typefinding work halfway stream,
- which doesn't actually work yet because id3demux doesn't implement
- _get_length().
+2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_loop):
- Fix interrupt event handling (#144436).
+2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_typefind):
- Hide unused glory.
+2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+ * 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).
- reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisenc.c: (raw_caps_factory):
- Fix weird caps (#157548).
+ Some minor docs fixes and additions; also add missing 'Since' bits.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_class_init):
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ Some minor docs fixes and additions; also add missing 'Since' bits.
-2004-11-06 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
+ Original commit message from CVS:
+ Patch by: Zeeshan Ali <zeenix gmail com>
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer),
+ (gst_base_rtp_audio_payload_push):
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ The recently-added gst_base_rtp_audio_payload_push() should take an
+ object of type GstBaseRTPAudioPayload as first argument (#431672).
- * gst/rtp/gstrtpgsmparse.c: (gst_rtpgsm_caps_nego):
- Add missing NULL terminator (#157543).
+2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/tcp/gsttcp.h:
- * gst/tcp/gsttcpclientsink.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpserversink.c:
- * gst/tcp/gsttcpserversrc.c:
- ports can go up to 65535. Move common defines to gsttcp.h
+2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ 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).
- * gst/videotestsrc/videotestsrc.c: (paint_setup_Y41B),
- (paint_hline_Y41B), (paint_setup_Y42B), (paint_hline_Y42B):
- Added two more colorspaces.
+2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-05 Wim Taymans <wim@fluendo.com>
+ 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/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (yuv422p_to_yuv422), (yuv420p_to_yuv422), (shrink12),
- (img_convert), (deinterlace_line), (deinterlace_line_inplace):
- More stride fixes.
+2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
-2004-11-05 Wim Taymans <wim@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).
- * gst/alpha/gstalpha.c: (gst_alpha_set_property), (gst_alpha_add),
- (gst_alpha_chroma_key), (gst_alpha_init_params), (gst_alpha_chain):
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_i420),
- (gst_videomixer_fill_checker), (gst_videomixer_blend_buffers),
- (gst_videomixer_loop):
- More stride fixes.
+2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-11-05 Benjamin Otte <otte@gnome.org>
+ tests/check/pipelines/.cvsignore: Fix build.
+ Original commit message from CVS:
+ * tests/check/pipelines/.cvsignore:
+ Fix build.
- * ext/mad/gstmad.c: (gst_mad_chain):
- don't overflow data buffer. Flush not needed sync data when syncing
- failed.
+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>
-2004-11-04 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_class_init), (gst_alpha_init), (gst_alpha_set_property),
- (gst_alpha_get_property), (gst_alpha_add), (gst_alpha_chroma_key),
- (gst_alpha_init_params), (gst_alpha_chain),
- (gst_alpha_change_state):
- Updated the chroma keying algorithm with something more
- sophisticated.
+2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ * gst-plugins-base.doap:
+ fix release date
+ Original commit message from CVS:
+ fix release date
- * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_i420),
- (gst_videomixer_fill_checker), (gst_videomixer_fill_color),
- (gst_videomixer_blend_buffers), (gst_videomixer_loop):
- Fix stride issues. Does not completely work for odd
- heights.
+2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/alpha/gstalpha.c: (gst_alpha_method_get_type),
- (gst_alpha_chroma_key), (gst_alpha_chain):
- Fix stride issues. Does not completely work for odd
- heights.
+2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-03 Christophe Fergeau <teuf@gnome.org>
+ 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.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- * gst/videoscale/videoscale.c: (videoscale_find_by_structure):
- leak fixes
+2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-11-03 Wim Taymans <wim@fluendo.com>
+ * gst/tcp/gstmultifdsink.c:
+ add debug
+ Original commit message from CVS:
+ add debug
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
- (gst_ffmpegcsp_avpicture_fill):
- * gst/ffmpegcolorspace/imgconvert.c: (avpicture_get_size),
- (avpicture_alloc):
- * gst/ffmpegcolorspace/imgconvert_template.h:
- Use correct _fill function to get correct strides.
-
-2004-11-02 David Schleef <ds@schleef.org>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree),
- (qtdemux_parse_udta), (qtdemux_tag_add), (gst_qtdemux_handle_esds):
- Change all g_print()s to debugging. Add a bunch of consistency
- checks.
-
-2004-11-02 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (try_to_link_1), (get_our_ghost_pad), (remove_element_chain),
- (unlinked), (no_more_pads), (close_link):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
- (unknown_type), (add_element_stream), (new_decoded_pad),
- (removed_decoded_pad), (setup_source):
- * gst/playback/gststreaminfo.c: (gst_stream_info_get_type),
- (gst_stream_info_class_init), (gst_stream_info_init),
- (gst_stream_info_new), (gst_stream_info_dispose),
- (stream_info_mute_pad), (gst_stream_info_set_property),
- (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Fix playback of multiple files.
- a slightly different approach to handling dynamic pad removals.
- This one only looks at pads that we have linked.
-
-2004-11-01 Christophe Fergeau <teuf@gnome.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_finalize): fix an "invalid
- free" warning from libc.
-
-2004-11-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (get_unconnected_element), (remove_starting_from), (pad_removed),
- (close_link):
- Implement support for dynamic pad changing. We listen to "live"
- pad removals (i.e. while playing) and re-setup autoplugging
- after that. Playbasebin/playbin need some more work for this
- to finally work, but decodebin supports (and replugs) chained
- ogg now.
-
-2004-11-02 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
- (gst_alsa_finalize):
- * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
- (gst_cdaudio_finalize):
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_finalize):
- * ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
- * ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
- (dvdreadsrc_finalize):
- * ext/flac/gstflacdec.c: (gst_flacdec_class_init),
- (gst_flacdec_finalize):
- * ext/flac/gstflacenc.c: (gst_flacenc_class_init),
- (gst_flacenc_finalize):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
- (gst_gnomevfssink_finalize):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
- (gst_gnomevfssrc_finalize):
- * ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
- (gst_fameenc_finalize):
- * ext/nas/nassink.c: (gst_nassink_class_init),
- (gst_nassink_finalize):
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
- (gst_sdlvideosink_class_init):
- * ext/sndfile/gstsf.c: (gst_sf_dispose):
- * gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
- * gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
- * gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
- * gst-libs/gst/xwindowlistener/xwindowlistener.c:
- (gst_x_window_listener_dispose):
- * gst/audioscale/gstaudioscale.c:
- * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
- (play_on_demand_finalize):
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
- * sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
- (cdplayer_finalize):
- * sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
- (gst_glimagesink_class_init):
- * sys/oss/gstosselement.c: (gst_osselement_class_init),
- (gst_osselement_finalize):
- * sys/oss/gstosssink.c: (gst_osssink_dispose):
- * sys/oss/gstosssrc.c: (gst_osssrc_dispose):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
- Fixes a bunch of problems with finalize and dispose functions,
- either assumptions that dispose is only called once, or not calling
- the parent class dispose/finalize function
-
-2004-11-01 Stefan Kost <ensonic@users.sf.net>
-
- * ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_link):
- added two api precondition guards
- use g_strdup with getenv to fix crash when using ENVVAR
-
-2004-11-01 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/esd/esdsink.c: (gst_esdsink_class_init),
- (gst_esdsink_finalize):
- Use a finalize function, not dispose, and more importantly,
- call the parent class finalize function too
-
-2004-11-01 Johan Dahlin <johan@gnome.org>
+2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c:
- * gst/tags/gstvorbistag.c:
- Plug leaks.
+ * 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
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- lotsa memleaks today. But they're all small...
+ * ext/ogg/gstoggmux.c:
+ * gst/gdp/gstgdpdepay.c:
+ debug changes
+ Original commit message from CVS:
+ debug changes
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- another memleak crushed
+ 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.
-2004-10-31 Benjamin Otte <otte@gnome.org>
+2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
- fix memleak
+ ext/theora/theoradec.c: Fix misleading comment.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_sink_event):
+ Fix misleading comment.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- Hack to prevent crash when going to READY inside signal handler
- while this function is active.
+ gst-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.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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-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.
-2004-10-31 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/playback/Makefile.am:
- We need the marshallers for decodebin, too.
+ 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.
-2004-10-30 David Schleef <ds@schleef.org>
+2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find): Make
- quicktime typefinding work with 64-bit offsets.
+ 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.
-2004-10-30 Jan Schmidt <thaytan@mad.scientist.com>
+2007-04-12 12:19:20 +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.
+ API: add gst_tag_freeform_string_to_utf8() (#405072).
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
+ API: add gst_tag_freeform_string_to_utf8() (#405072).
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
+ Use gst_tag_freeform_string_to_utf8() here.
-2004-10-29 Martin Pitt <martin.pitt@canonical.com>
+2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/tcp/gstmultifdsink.c:
+ log tweaking
+ Original commit message from CVS:
+ log tweaking
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
- Don't touch buffer if it is of size 0 (fixes #151064).
+2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
+ Original commit message from CVS:
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event):
+ Make sure we set the IN_CAPS flag correctly.
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
+ Get the IN_CAPS flag before we call functions that mess with the flags.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push), (gst_ogg_pad_push):
- Synchronized discont handling.
+2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_push):
- Make seeking sort-of exact again (fixes #156387).
+2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/gdp/gstgdpdepay.c:
+ * gst/gdp/gstgdppay.c:
+ adding debugging
+ Original commit message from CVS:
+ adding debugging
- * gst/playback/gstplaybasebin.c: (unknown_type),
- (add_element_stream), (new_decoded_pad),
- (gst_play_base_bin_change_state):
- * gst/playback/gststreaminfo.c: (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Make caps explicitely available. Makes testing for unsupported
- types possible. Improves error reporting.
+2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * common:
+ * gst-plugins-base.spec.in:
+ update spec file for RTP changes
+ Original commit message from CVS:
+ update spec file for RTP changes
- * gst/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_buffer_to_default_format):
- Really don't touch read-only buffers (#156563).
+2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-29 Sebastien Cote <sc5@hermes.usherb.ca>
+ 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.
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak (#155223).
+ tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
+ Original commit message from CVS:
+ * tests/check/elements/playbin.c:
+ (test_sink_usage_video_only_stream), (playbin_suite):
+ Add small test for stream-info-value-array code paths.
-2004-10-29 Wim Taymans <wim@fluendo.com>
+2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/tcp/.cvsignore:
- * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
- (gst_multifdsink_class_init), (gst_multifdsink_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_remove_client_link), (is_sync_frame),
- (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Added burst on connect sync_method, deprecated sync_clients,
- streamlined the sync code some more.
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_skew_slaving):
+ Don't try to create invalid calibration parameters by making the
+ internal time go backwards, instead make external time go forward.
-2004-10-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * gst/playback/gstplaybasebin.c: (thread_error), (setup_source),
- (gst_play_base_bin_change_state):
- Improve error reporting.
+ gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst/playback/gstplaybasebin.c: (add_stream):
+ Fix leak in add_stream(), when g_value_set_object() increases the
+ refcount of streaminfo object. Fixes #426250.
-2004-10-28 Wim Taymans <wim@fluendo.com>
+2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
- * gst/tcp/Makefile.am:
- * gst/tcp/fdsetstress.c: (mess_some_more), (run_test), (main):
- * gst/tcp/gstfdset.c: (nearest_pow), (resize), (ensure_size),
- (gst_fdset_new), (gst_fdset_free), (gst_fdset_set_mode),
- (gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- Added more locks around fdset structures. Fixed/reworked
- the poll array resizing code.
- Added stress test for fdset.
-
-2004-10-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
- fix build
-
-2004-10-28 Benjamin Otte <otte@gnome.org>
-
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
- fix link function to always query channels and query width for
- floats
- * configure.ac:
- add equalizer dir
- * gst/equalizer/Makefile.am:
- * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
- (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
- (gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
- (arg_to_scale), (setup_filter),
- (gst_iir_equalizer_compute_frequencies),
- (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
- (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
- (plugin_init):
- add an equalizer
-
-2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
-
- Submitted by: Kjartan Maraas <kmaraas@broadpark.no>
+ gst/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.
- * po/LINGUAS:
- * po/nb.po:
- Added Norwegian Bokmaal translation
+2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
-2004-10-27 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
+ (deactivate_free_recursive):
+ Decodebin2 doesn't unref pads it obtains in some occasions:
+ - multiqueue src pads, when either connecting further or exposing
+ - sink pads of new autoplugged elements
+ - peer pads when recursively freeing elements
+ Fixes #425455.
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- Don't break on options (fixes #156488).
+2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- * ext/cdaudio/Makefile.am:
- * sys/Makefile.am:
- fix build on older automake
+2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-26 Wim Taymans <wim@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.
- * ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_push), (gst_dvdec_loop):
- Allow a little margin when negotiating the framerate.
+2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
+
+ with some minor changes
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler dot de>
+ with some minor changes
+ * gst-libs/gst/floatcast/floatcast.h:
+ Use more efficient float endianness conversion functions that don't
+ involve 2 function calls per value.
+ * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
+ (check_default), (audio_convert_prepare_context):
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_parse_caps), (make_lossless_changes):
+ Support non-native endianness floats as input and output.
+ Fixes #339838.
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST):
+ Add unit tests for the non-native endianness float conversions.
+
+2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_base_init),
+ (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
+ (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_set_gst_timestamp),
+ (gst_base_rtp_depayload_change_state),
+ (gst_base_rtp_depayload_set_property),
+ (gst_base_rtp_depayload_get_property):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Add Private structure.
+ Bring element code to 2007.
+ Parse clock-base caps param and use it when generating the
+ newsegment.
+ Reset variables before going to PAUSED.
+ Fix some docs.
+
+2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Add RTCP docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_get_adapter):
+ Add RTCP docs.
+ Fix some more docs.
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
+ (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
+ (gst_rtcp_buffer_get_packet_count), (read_packet_header),
+ (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
+ (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
+ (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
+ (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
+ (gst_rtcp_packet_sr_get_sender_info),
+ (gst_rtcp_packet_sr_set_sender_info),
+ (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
+ (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
+ (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
+ (gst_rtcp_packet_sdes_get_chunk_count),
+ (gst_rtcp_packet_sdes_first_chunk),
+ (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
+ (gst_rtcp_packet_sdes_first_item),
+ (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
+ (gst_rtcp_packet_bye_get_ssrc_count),
+ (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
+ (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
+ (gst_rtcp_packet_bye_get_reason_len),
+ (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
+ * gst-libs/gst/rtp/gstrtcpbuffer.h:
+ Add new helper object for parsing and creating RTCP messages.
+
+2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ PCM samples with width=8 must be always unsigned, no matter what
+ depth they have.
+
+2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
+
+ gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
+ Original commit message from CVS:
+ 2007-03-29 Andy Wingo <wingo@pobox.com>
+ * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
+ perfect offsets also, not just timestamps.
+ * tests/check/elements/videorate.c (test_more): Test that given
+ any incoming offsets, that videorate produces perfect offsets.
+
+2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add some more RIFF formats.
+
+2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ (gst_rtp_buffer_default_clock_rate):
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Fix fixed payload names and docs.
+ Added method to get the default clock rates of fixed payload types.
+ API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
+
+2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
+
+ tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
+ Original commit message from CVS:
+ * tests/check/pipelines/.cvsignore:
+ Add new vorbisdec test to cvsignore.
+
+2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
+ (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
+ (gst_base_audio_sink_set_property),
+ (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
+ (clock_convert_external), (gst_base_audio_sink_resample_slaving),
+ (gst_base_audio_sink_skew_slaving),
+ (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
+ (gst_base_audio_sink_async_play):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Store private stuff in GstBaseAudioSinkPrivate.
+ Add configurable clock slaving modes property.
+ API:: GstBaseAudioSink::slave-method property
+ Some more latency reporting tweaks.
+ Added skew based clock slaving correction and make it the default until
+ the resampling method is more robust.
+
+2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
+
+ gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c:
+ Add docs to the integer pack functions and implement proper
+ rounding. Before we had rounding towards negative infinity, i.e.
+ always the smaller number was taken. Now we use natural rounding,
+ i.e. rounding to the nearest integer and to the one with the largest
+ absolute value for X.5. The old rounding introduced some minor
+ distortions. Fixes #420079
+ * tests/check/elements/audioconvert.c: (GST_START_TEST):
+ Fix one unit test that assumed the old rounding and added unit tests
+ for checking signed/unsigned int16 <-> signed/unsigned int16 with
+ depth 8, one for signed int16 <-> unsigned int16 and one for the new
+ rounding from signed int32 to signed/unsigned int16.
+
+2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
+ (gst_audio_convert_transform_caps):
+ Fix typo in debug line introduced recently, as pointed out on irc.
-2004-10-26 Stefan Kost <ensonic@users.sf.net>
+2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/level/gstlevel.c:
- synchonised naming of pads and pad-templates
+ Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Make sure we parse floating-point numbers in vorbis comments
+ correctly with either '.' or ',' as separator, no matter what
+ the current locale is. Add unit test for this too.
-2004-10-26 Wim Taymans <wim@fluendo.com>
+2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * tests/check/pipelines/vorbisdec.c:
+ commit new file
+ Original commit message from CVS:
+ commit new file
-2004-10-26 Iain <iaingnome@gmail.com>
+2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
- * 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.
+ gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
+ Original commit message from CVS:
+ Patch by: René Stadler <mail at renestadler de>
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
+ When writing out floating-point numbers to vorbis comment tags, always
+ use the same character as separator no matter what the current locale is
+ (fixes #423051).
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ Add unit tests for replaygain tags in vorbis comments (closes #423055).
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
+ vorbis_handle_data_packet):
+ Correctly set DURATION to generate a timestamp-continuous stream.
+ One bug left at the end; see
+ ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
+ Add a test to check this. Without the above patch this test fails.
-2004-10-26 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_init),
- (gst_gnomevfssink_handle_event), (gst_gnomevfssink_chain):
- Implement seeking.
+ gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/Makefile.am:
+ The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
-2004-10-25 James Henstridge <james@jamesh.id.au>
+2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
- Reviewed by: David Schleef <ds@schleef.org>
+ * gst-plugins-base.spec.in:
+ update spec file
+ Original commit message from CVS:
+ update spec file
- * 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.
+2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
-2004-10-25 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- reviewed by: Ronald Bultje <rbultje at gnome dot org>
+2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
- Fix for some v4l cards which hang in v4lsrc
+ 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.
-2004-10-25 Wim Taymans <wim@fluendo.com>
+2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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-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.
-2004-10-25 Wim Taymans <wim@fluendo.com>
+2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/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.
-2004-10-25 Takao Fujiwara <Takao.Fujiwara@Sun.COM>
+2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
- Reviewd by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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).
+2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
+ If we get a zero-sized input buffer, don't pass it to libvorbis, as
+ that marks EOS internally. After that, libvorbis will buffer all
+ input data, and encode none of it, eventually leading to memory
+ exhaustion.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Export bitrate as metadata.
+2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
+ Original commit message from CVS:
+ * REQUIREMENTS: Update this file, change the formatting to make
+ it more consistent, plus more machine readable.
- * ext/speex/gstspeexdec.c: (gst_speex_dec_init),
- (speex_dec_convert):
- sinkconvert function so oggdemux can get the file length (totem).
+2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
-2004-10-25 James Morrison <ja2morri@csclub.uwaterloo.ca>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
- * 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?
+ 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.
-2004-10-25 Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * sys/oss/gstosssrc.c: (gst_osssrc_get_time), (gst_osssrc_get),
- (gst_osssrc_src_query):
- * sys/oss/gstosssrc.h:
- OK, so people want offset in DEFAULT. This time, actually fix all
- cases.
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Add FPS properly.
+2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/gdp/gstgdpdepay.c:
+ add buffer logging
+ Original commit message from CVS:
+ add buffer logging
- * gst/asfdemux/gstasfmux.c:
- * gst/avi/gstavimux.c:
- Framerate.
+2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_set_property):
- Fix properties (channel, norm, frequency).
+2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * gst/audioresample/gstaudioresample.c:
+ add debugging and reformat docs
+ Original commit message from CVS:
+ add debugging and reformat docs
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_get_property):
- Flag typo.
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_set_defaults):
- No warnings.
+2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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...
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_clear_format_list):
- Fix hang.
+2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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.
- * 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.
+2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.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).
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_getcaps):
- Quiet.
+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).
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Don't mix bytes and samples.
+ 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.
-2004-10-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * 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>
+2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
- * ext/Makefile.am:
- Fix the build fixes.
+ gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
+ Replace AC3 typefinder with one that isn't terrible, and actually
+ works usefully.
-2004-10-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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/audioconvert/gstaudioconvert.c: fix error category and translatable string
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c:
+ (gst_audio_convert_transform):
+ fix error category and translatable string
-2004-10-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ pkgconfig/: Fix up utils => pbutils here too.
+ Original commit message from CVS:
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ Fix up utils => pbutils here too.
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- We love it if files that start at zero work too...
+ gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (handle_buffer):
+ Break out of loop in chain function as soon as possible if we get
+ a non-OK flow return.
-2004-10-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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/gstoggdemux.c: (gst_ogg_demux_iterate):
- Handle files with missing EOS headers.
+2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-10-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ ext/ogg/gstoggdemux.c: Fix confusing debug message.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
+ Fix confusing debug message.
- * gst/tcp/gsttcpserversink.c:
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_init_send):
- Zero some variables first (need for accept not to return EINVAL)
+2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_push):
- * ext/theora/theoradec.c: (theora_dec_sink_convert),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (gst_vorbis_dec_init), (vorbis_dec_convert), (vorbis_dec_chain):
- Seeking and querying finetune.
+2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/tcp/gstmultifdsink.c:
+ update docs
+ Original commit message from CVS:
+ update docs
- * configure.ac:
- * ext/Makefile.am:
- * ext/raw1394/Makefile.am:
- fix the build
+2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ configure.ac: Back to CVS
+ Original commit message from CVS:
+ * configure.ac:
+ Back to CVS
- * 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.
+=== release 0.10.12 ===
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event):
- Silence.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-decodebin2.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ Release 0.10.12
+ Original commit message from CVS:
+ Release 0.10.12
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-07 15:35:26 +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),
- (gst_ogg_pad_push):
- Yay for non-lineair granulepos in theora.
+ * 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
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ 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
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_push):
- Start at zero.
- * ext/theora/theoradec.c: (theora_dec_chain):
- Skip headers. Bad idea for chained ogg, but fixes seeking.
+ gst-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.
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * configure.ac:
- I swear, this is the last time I touch this.
+ 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
-2004-10-18 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_pad_populate),
- (_read_bos_process), (gst_ogg_demux_iterate), (gst_ogg_pad_new):
- Faster seeking.
- * ext/theora/theoradec.c: (theora_dec_sink_convert):
- Time-to-default conversion.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Don't error on unknown packets, just skip. We should probably
- read them if we want to support chained ogg.
+ 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.
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Added cdaudio to wrong list.
+ 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.
-2004-10-18 Wim Taymans <wim@fluendo.com>
+2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- Revive cdaudio.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_push):
- * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
- (gst_smokeenc_resync), (gst_smokeenc_chain):
- Fix mimetype on smoke encoder.
- Add aspect ratio to dvdec. Not sure if these
- values are correct though....
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_class_init):
- Fix vorbis property descriptions and ranges.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate):
- Really do nothing when no data is available.
- Go to the playing state when the stream is not seekable
- instead of failing.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * ext/cdaudio/gstcdaudio.c: (_do_init), (gst_cdaudio_base_init),
- (gst_cdaudio_get_event_masks), (gst_cdaudio_send_event),
- (gst_cdaudio_query), (plugin_init), (cdaudio_uri_get_type),
- (cdaudio_uri_get_protocols), (cdaudio_uri_get_uri),
- (cdaudio_uri_set_uri), (cdaudio_uri_handler_init):
- Added uri handler for cd://
- Port to new API.
-
-2004-10-18 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
- (gst_decode_bin_init), (find_compatibles), (close_pad_link),
- (try_to_link_1), (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (gen_preroll_element),
- (remove_prerolls), (unknown_type), (add_element_stream),
- (new_decoded_pad), (setup_source), (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_link_stream):
- * gst/playback/gstplaybin.c: (gen_video_element),
- (gen_vis_element), (remove_sinks), (setup_sinks):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (gst_stream_info_get_type), (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (stream_info_mute_pad),
- (gst_stream_info_set_property), (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- Add sink padtemplate to decodebin.
- Added some more comments.
- Make queue size configurable in playbasebin.
- Added possibility to use elements as sinks (ex cdaudio).
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/speex/gstspeexenc.c: (gst_speexenc_class_init),
- (gst_speexenc_chain):
- Fix speex timestamps so that it gets muxed properly.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_get_type),
- (gst_dv1394src_base_init), (gst_dv1394src_class_init),
- (gst_dv1394src_init), (gst_dv1394src_dispose),
- (gst_dv1394src_iso_receive), (gst_dv1394src_discover_avc_node),
- (gst_dv1394src_change_state), (gst_dv1394src_get_event_mask),
- (gst_dv1394src_event), (gst_dv1394src_get_formats),
- (gst_dv1394src_convert), (gst_dv1394src_get_query_types),
- (gst_dv1394src_query), (gst_dv1394src_uri_get_type),
- (gst_dv1394src_uri_get_protocols), (gst_dv1394src_uri_get_uri),
- (gst_dv1394src_uri_set_uri), (gst_dv1394src_uri_handler_init):
- * ext/raw1394/gstdv1394src.h:
- Added conversion/query functions.
- Update buffer timestamps,
- Added signals.
- Added uri dv:// so it might play from the firewire in playbin.
- Fix a possible leak.
- Added debugging.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
- (gst_dv1394src_init), (gst_dv1394src_set_property),
- (gst_dv1394src_get_property), (gst_dv1394src_iso_receive),
- (gst_dv1394src_discover_avc_node), (gst_dv1394src_change_state):
- * ext/raw1394/gstdv1394src.h:
- Added AV/C VTR control support needed for some cameras.
- Added automatic port detection.
- Added properties for selecting the channel.
- The configure.ac script is not yet updated to reflect the
- new libavc1394 and librom1394 dependencies.
-
-2004-10-15 Wim Taymans <wim@fluendo.com>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse), (gst_qtdemux_handle_esds):
- An esds box is not a container.
- Fix parsing of mp4v boxes.
- Do not try to renegotiate fps for each frame. Need to
- find a better method. This should fix mp4 playback.
-
-2004-10-14 David Schleef <ds@schleef.org>
-
- * configure.ac: update for swfdec-0.3 and liboil-0.2
- * ext/swfdec/gstswfdec.c: update for swfdec-0.3
- * ext/swfdec/gstswfdec.h: same
- * gst/videofilter/gstvideobalance.c: update for liboil-0.2
- * gst/videotestsrc/videotestsrc.c: same
-
-2004-10-14 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
- (is_sync_frame), (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
- Turn warnings into info.
- Don't allow a state change in the streaming thread.
-
-2004-10-14 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/oggvorbisenc.c:
- * ext/vorbis/vorbisdec.c:
- fix template sample rate
+ * tests/check/libs/.gitignore:
+ moap ignore
+ Original commit message from CVS:
+ moap ignore
-2004-10-13 Wim Taymans <wim@fluendo.com>
+2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset), (gst_mad_chain):
- Decoding the header first fixes some problems in resyncing
- in more mp3s.
+ * win32/common/config.h:
+ * win32/common/libgstutils.def:
+ update defs
+ Original commit message from CVS:
+ update defs
-2004-10-12 Wim Taymans <wim@fluendo.com>
+2007-03-04 23:39:51 +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.
+ 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
-2004-10-12 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get),
- (gst_gnomevfssrc_srcpad_query), (gst_gnomevfssrc_srcpad_event):
- Some debug.
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
- (gst_avi_demux_handle_src_event), (gst_avi_demux_read_superindex),
- (gst_avi_demux_read_subindexes), (gst_avi_demux_add_stream),
- (gst_avi_demux_stream_index), (gst_avi_demux_skip),
- (gst_avi_demux_sync), (gst_avi_demux_stream_scan),
- (gst_avi_demux_massage_index), (gst_avi_demux_stream_header):
- * gst/avi/gstavidemux.h:
- Support for openDML-2.0 indx/ix## chunks. Support for broken index
- recovery (where, if part of the index is broken, we will still read
- the rest of the index and recover the broken part by stream
- scanning). More broken media support. EOS workarounds. General AVI
- braindamage headache recovery. Aspirin included.
+2007-02-28 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>
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * tests/check/generic/states.c:
+ plug test leak
+ Original commit message from CVS:
+ plug test leak
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_open),
- (cdparanoia_event), (cdparanoia_query):
- Get rid of hideous lead-in.
+2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-11 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
- * gst/playback/gstplaybasebin.c: (setup_source):
- Wrong var used to get g_list_next.
+2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-11 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
- (cdparanoia_get), (cdparanoia_open):
- Report discid as metadata, add duration.
+2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-10-11 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Cleanup the previous pipeline a little earlier for the
- case that a source element provides raw data.
+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.
-2004-10-11 Benjamin Otte <otte@gnome.org>
+2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
- * 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.
+ 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).
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c:
+ Improve docs: point out that the application needs to assist playbin
+ with buffering.
-2004-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_src_link), (gst_v4lsrc_getcaps):
- Fix for webcams that support only specific width or height
+ Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
+ Original commit message from CVS:
+ * gst-libs/gst/utils/install-plugins.c:
+ * gst-libs/gst/utils/missing-plugins.c:
+ * tests/check/libs/utils.c: (missing_msg_check_getters):
+ Change GStreamer marker prefix in detail string from 'gstreamer.net'
+ to just 'gstreamer'. Document the caps string component of the
+ decoder/encoder detail a bit better, since not everyone will be
+ familiar with the GStreamer media type/caps system (but they better
+ enjoy nested itemized lists).
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
- Fix wrong discont event setup (fixes #154967).
+2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
-2004-10-09 Sebastien Cote <sc5@hermes.usherb.ca>
+ * 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
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Error out on invalid data (fixes #154807).
+ 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.
-2004-10-09 Tim-Philipp Müller <t.i.m@zen.co.uk>
+2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.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.
- * ext/dvdread/dvdreadsrc.c: (_read):
- Make titles > 0 work again (fixes #154834).
+2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.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-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_template_caps):
- WMV3 missing in template caps.
+2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.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/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- OK, so the original code was too strict. It makes random AVI files
- hang for seconds upon opening, which is unacceptable and is far
- beyond the original goal of getting multiple chunks for one-chunk
- sounc stream files. So now do just that.
+2007-02-20 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).
-2004-10-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ gst-libs/gst/audio/audio.c: Fix documentation.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.c:
+ Fix documentation.
-2004-10-08 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
- * 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.
+ gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
+ Original commit message from CVS:
+ Patch by: Yves Lefebvre <ivanohe abacom com>
+ * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
+ Don't leak caps. Fixes #408278.
-2004-10-08 Wim Taymans <wim@fluendo.com>
+2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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"...
+ More docs coverage and some ChangeLog surgery (add missing names)
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ * ext/ogg/gstoggdemux.h:
+ * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
+ (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
+ (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
+ * gst-libs/gst/audio/audio.h:
+ * gst-libs/gst/audio/gstaudiofilter.h:
+ * gst-libs/gst/interfaces/videoorientation.h:
+ * gst/adder/gstadder.h:
+ More docs coverage and some ChangeLog surgery (add missing names)
-2004-10-08 Wim Taymans <wim@fluendo.com>
+2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2004-10-08 Milosz Derezynski <internalerror.rez@fhtw-berlin.de>
+2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/cdparanoia/gstcdparanoia.c: (_do_init),
- Added reworked patch from #154903 from milosz derezynski (deadchip).
+ 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>
-2004-10-08 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
- * ext/cdparanoia/gstcdparanoia.c: (_do_init),
- (cdparanoia_base_init), (cdparanoia_class_init), (cdparanoia_init),
- (cdparanoia_dispose), (cdparanoia_get), (cdparanoia_change_state),
- (cdparanoia_convert), (cdparanoia_uri_get_type),
- (cdparanoia_uri_get_protocols), (cdparanoia_uri_get_uri),
- (cdparanoia_uri_set_uri), (cdparanoia_uri_handler_init):
- * ext/cdparanoia/gstcdparanoia.h:
- This adds the cdda://<tracknum> uri.
+2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-08 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/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.
+2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-08 Julien MOUTTE <julien@moutte.net>
+ 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.
- * 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.
+2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-07 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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.
+2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
-2004-10-07 Wim Taymans <wim@fluendo.com>
+ 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).
- * 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.
+2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-07 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/typefind/gsttypefindfunctions.c: (ac3_type_find),
- (plugin_init):
- Added lame audio/x-ac3 typefind function.
+2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
+ Use gst_gdouble_to_guint64 for conversions.
+ * win32/common/config.h.in:
+ Add a define for GST_INSTALL_PLUGINS_HELPER
+ * win32/common/libgstaudio.def:
+ * win32/common/libgstcdda.def:
+ * win32/common/libgstnetbuffer.def:
+ * win32/common/libgstrtp.def:
+ * win32/common/libgutils.def:
+ Add new exported functions.
+ * win32/vs6/gst_plugins_base.dsw:
+ * win32/vs6/libgstdecodebin.dsp:
+ * win32/vs6/libgstnetbuffer.dsp:
+ * win32/vs6/libgstplaybin.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgstvorbis.dsp:
+ * win32/vs6/libgstcdda.dsp:
+ * win32/vs6/libgstgdp.dsp:
+ * win32/vs6/libgstutils.dsp:
+ Update and add new project files.
- * configure.ac:
- bump nano to cvs
+2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.8.5 ===
+ 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.
-2004-10-06 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS:
- * RELEASE:
- * configure.ac:
- releasing 0.8.5, "Take You On"
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
+
+2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (add_fakesink),
+ (gst_decode_bin_change_state):
+ * gst/playback/gstdecodebin2.c: (add_fakesink),
+ (gst_decode_bin_change_state):
+ Don't error out if there is no fakesink in the READY to NULL state
+ change, since when decodebin is re-used, we're only adding the
+ fakesink element in READY to PAUSED.
+ * tests/check/elements/decodebin.c:
+ (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
+ (decodebin_suite):
+ Minimal unit test to make sure we can use the same decodebin
+ instance twice (at least with audiotestsrc input).
-2004-10-06 Wim Taymans <wim@fluendo.com>
+2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (no_more_pads), (close_link), (type_found):
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- * gst/playback/gstplaybin.c: (gen_video_element):
- Do not signal the no_more_pads after the first pad when
- we are plugging a non dynamic element with multiple
- output pads (like swfdec, dvdec, ...).
+ ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
+ Original commit message from CVS:
+ * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
+ Try to get devic-name from device string first, and from handle only
+ as fallback (seems to yield better results and is more robust
+ against buggy probing code on the application side).
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
- * configure.ac:
- bump for prerelease
+ ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
+ Original commit message from CVS:
+ Based on patch by: Julien Puydt <julien.puydt at laposte net>
+ * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
+ (gst_alsa_find_device_name):
+ * ext/alsa/gstalsa.h:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
+ Improve device-name detection a bit, especially in the case where
+ the device is not actually open (#405020, #405024). Move common code
+ into gstalsa.c instead of duplicating it.
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c:
- add ATRAC3 to STATIC CAPS to fix a warning
+ 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.
- * gst/matroska/ebml-read.c:
- * gst-libs/gst/riff/riff-read.c:
- fix typos
+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.
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c:
- generate caps for ATRAC3 audio streams
+ 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/realmedia/rmdemux.c:
- generate caps for ATRAC3 audio streams
+2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ * 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/wavparse/Makefile.am
- * gst/wavparse/riff.h
- * gst/wavparse/wavparse.vcproj
- riff.h removal (unused and duplication with riff-ids.h)
+2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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/wavparse/gstwavparse.h
- remove duplicated defines for audio codec codes
+2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-ids.h
- * gst/wavenc/riff.h:
- add "4CC" code for ATRAC3 audio streams
- add "4CC" code for ITU_G721_ADPCM (unused for now)
+ 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.
-2004-10-06 Wim Taymans <wim@fluendo.com>
+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/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_loop):
- Actually _do_ negotiation. Pass gdouble as arg instead
- of guint64 for the framerate.
+2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Wim Taymans <wim@fluendo.com>
+ 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/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.
+2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- written by: Gora Mohanty <gora_mohanty@yahoo.co.in>
+2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/LINGUAS:
- * po/or.po:
- add Oriya translation
+ tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
+ Original commit message from CVS:
+ * tests/check/libs/utils.c:
+ (test_base_utils_install_plugins_do_callout):
+ Don't hard-code temp directory for test helper; use GLib functions
+ to write out file and do error checking etc.
-2004-10-05 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
+ Original commit message from CVS:
+ * gst-libs/gst/utils/Makefile.am:
+ * gst-libs/gst/utils/base-utils.h:
+ * gst-libs/gst/utils/install-plugins.c:
+ (gst_install_plugins_context_set_xid),
+ (gst_install_plugins_context_new),
+ (gst_install_plugins_context_free),
+ (gst_install_plugins_get_helper),
+ (gst_install_plugins_spawn_child),
+ (gst_install_plugins_return_from_status),
+ (gst_install_plugins_installer_exited),
+ (gst_install_plugins_async), (gst_install_plugins_sync),
+ (gst_install_plugins_return_get_name),
+ (gst_install_plugins_installation_in_progress):
+ * gst-libs/gst/utils/install-plugins.h:
+ API: add API for applications to initiate installation of missing
+ plugins, ie. gst_install_plugins_async() primarily.
+ Based on libgimme-codec by Ryan Lortie.
+ * configure.ac:
+ Add --with-install-plugins-helper configure option so distros can specify
+ the path of the helper script or program to call when plugin installation
+ is requested (distros: please do any argument munging in this helper
+ script instead of patching GStreamer to pass arguments differently
+ to another program directly).
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ Build and document new API.
+ * tests/check/libs/utils.c: (result_cb),
+ (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
+ (libgstbaseutils_suite):
+ Some simple checks for the new API.
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- Prevent overwrite of size member. Makes audio sound crappy.
+2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-05 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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/typefind/gsttypefindfunctions.c: (plugin_init):
- Add rmvb to the list of known RealMedia extensions
+2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-05 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float), (double),
+ (audio_convert_get_func_index):
+ * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
+ (make_lossless_changes):
+ Support for 64-bit float audio in audioconvert (#339837)
- * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
- There was something wrong with the index massaging.
+2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
-2004-10-04 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/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.
+2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-10-04 Wim Taymans <wim@fluendo.com>
+ 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
- * 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>
+2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- you need at least 1.0.4 of speex
+ 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.
-2004-10-04 Iain <iaingnome@gmail.com>
+2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/speex/gstspeexdec.h: Revert the includes changes.
+ 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/speex/gstspeexenc.[ch]: Revert the includes changes.
+2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
-2004-09-30 Iain <iaingnome@gmail.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.
- * sys/sunaudio/gstsunaudio.c (gst_sunaudiosink_open): Use the device
- found during init or set as a property instead of hardcoding /dev/audio
+2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/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.
+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.
-2004-10-04 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
- (gst_avi_demux_stream_scan), (sort), (gst_avi_demux_massage_index),
- (gst_avi_demux_stream_header), (gst_avi_demux_stream_data):
- Improve allocation, cutting and sorting of the index. How takes a
- few seconds instead of minutes.
+ gst/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.
-2004-10-03 Christophe Fergeau <teuf@gnome.org>
+2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/realmedia/rmdemux.c: (gst_rmdemux_parse_mdpr):
- fixed compilation
+ 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.
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ 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).
-2004-10-02 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_video_template_caps):
- Add 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.
+ 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().
-2004-10-01 Christian Schaller <christian@fluendo.com>
+2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ivorbis/vorbis.c:
- Patch from Phil Blundell (Bug 152341)
+ 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.
-2004-10-01 Wim Taymans <wim@fluendo.com>
+2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.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.
+ 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.
-2004-10-01 Wim Taymans <wim@fluendo.com>
+2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
- (gst_wavparse_stream_init), (gst_wavparse_fmt),
- (gst_wavparse_other), (gst_wavparse_loop),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query),
- (gst_wavparse_srcpad_event):
- * gst/wavparse/gstwavparse.h:
- Added some more debugging info.
- Fix the case where the length of the file is 0.
- Make sure we seek to sample borders.
+ 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.
-2004-10-01 Wim Taymans <wim@fluendo.com>
+2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.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
+ 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.
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dvdnav/dvdnavsrc.c: (dvdnav_handle_navigation_event):
- Don't use g_print(); use GST_DEBUG().
+ 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).
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_queue_pads):
- Handle EOS properly.
+ 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.
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_sinkconnect),
- (gst_faad_chain), (gst_faad_change_state):
- * ext/faad/gstfaad.h:
- Allow playback of raw (unframed) MPEG AAC files (#148993).
+ gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ On second thought, use "depth" field rather than "bpp" field.
-2004-10-01 Sebastien Cote <sc5@hermes.usherb.ca>
+2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Throw error if we didn't recognize the stream. Fixes #152289.
+2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_link):
- Fix negotiation.
+2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Francis Labonte <francis_labonte@hotmail.com>
+ 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.
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
- Fix memleak.
+ gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
+ Lower probability a bit if the marker isn't right at the start,
+ to decrease the chance of false positives.
-2004-10-01 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
+2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * sys/sunaudio/gstsunaudio.c: (gst_sunaudiosink_setparams):
- Solve #152805.
- * sys/sunaudio/gstsunmixer.c: (gst_sunaudiomixer_set_mute):
- Solve 152806.
+2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_audio_caps_with_data):
- Add codec_data handling (like asfdemux used to do).
- * gst/asfdemux/gstasf.c: (plugin_init):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
- Use riff-media for caps creation instead of our own (mostly
- broken) copy of its functions.
-
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_try_capture):
- Don't actually error out if we get another return value than
- -EINVAL. Opposite to what I first thought, drivers have random
- return values for this, although -EINVAL is the expected return
- value. Since this is not fatal, we shouldn't use
- GST_ELEMENT_ERROR() but just GST_ERROR_OBJECT().
-
-2004-10-01 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
- (dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
- (dvdreadsrc_get_property), (_open), (_seek), (_read),
- (dvdreadsrc_get), (dvdreadsrc_open_file),
- (dvdreadsrc_change_state):
- Fix. Don't do one big huge loop around the whole DVD, that will
- cache all data and thus eat sizeof(dvd) (several GB) before we
- see something.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
- Actually NULL'ify event after using it.
- * gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
- (gst_ebml_read_handle_event), (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_skip):
- Handle events.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
- (gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
- (gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
- Fix timing (this will probably break if I seek using menus, but
- I didn't get there yet). VOBs and normal DVDs should now work.
- Add a mpeg2-only pad with high rank so this get autoplugged for
- MPEG-2 movies.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
- (gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
- (gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
- (gst_mpeg_demux_get_audio_stream),
- (gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
- (gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
- Use this as second rank for MPEG-1 and MPEG-2. Still use this for
- MPEG-1 but use dvddemux for MPEG-2.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
- (gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
- (gst_mpeg_parse_parse_packhead):
- Timing. Only add pad template if it exists. Add sink template from
- class and not from ourselves. This means we will always use the
- correct sink template even if it is not the one defined in this
- file.
-
-2004-09-29 Wim Taymans <wim@fluendo.com>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_packet),
- (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
- Fix playback of mpeg again, timestamps where screwed up by
- patch 1.61.
-
-2004-09-29 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_src_query):
- Only return true if we actually filled something in. Prevents
- player applications from showing a random length for flac files.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
- (gst_riff_read_use_event), (gst_riff_read_handle_event),
- (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
- (gst_riff_read_strf_vids_with_data),
- (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
- OK, ok, so I implemented event handling. Apparently it's normal
- that we receive random events at random points without asking
- for it.
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset),
- (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
- (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
- (gst_avi_demux_sync), (gst_avi_demux_stream_scan),
- (gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
- (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
- (gst_avi_demux_stream_data), (gst_avi_demux_loop):
- * gst/avi/gstavidemux.h:
- Implement non-lineair chunk handling and subchunk processing.
- The first solves playback of AVI files where the audio and video
- data of individual buffers that we read are not synchronized.
- This should not happen according to the wonderful AVI specs, but
- of course it does happen in reality. It is also a prerequisite for
- the second. Subchunk processing allows us to cut chunks in small
- pieces and process each of these pieces separately. This is
- required because I've seen several AVI files with incredibly large
- audio chunks, even some files with only one audio chunk for the
- whole file. This allows for proper playback including seeking.
- This patch is supposed to fix all AVI A/V sync issues.
- * gst/flx/gstflxdec.c: (gst_flxdec_class_init),
- (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
- Work.
- * gst/modplug/gstmodplug.cc:
- Proper return value setting for the query() function.
- * gst/playback/gstplaybasebin.c: (setup_source):
- Being in non-playing state (after, e.g., EOS) is not necessarily
- a bad thing. Allow for that. This fixes playback of short files.
- They don't actually playback fully now, because the clock already
- runs. This means that small files (<500kB) with a small length
- (<2sec) will still not or barely play. Other files, such as mod
- or flx, will work correctly, however.
-
-2004-09-28 Wim Taymans <wim@fluendo.com>
-
- * ext/speex/gstspeex.c: (plugin_init):
- * ext/speex/gstspeexdec.c: (gst_speex_dec_base_init),
- (gst_speex_dec_class_init), (speex_dec_get_formats),
- (speex_get_event_masks), (speex_get_query_types),
- (gst_speex_dec_init), (speex_dec_convert), (speex_dec_src_query),
- (speex_dec_src_event), (speex_dec_event), (speex_dec_chain),
- (gst_speexdec_get_property), (gst_speexdec_set_property),
- (speex_dec_change_state):
- * ext/speex/gstspeexdec.h:
- * ext/speex/gstspeexenc.c: (gst_speexenc_get_formats),
- (gst_speexenc_get_type), (speex_caps_factory), (raw_caps_factory),
- (gst_speexenc_base_init), (gst_speexenc_class_init),
- (gst_speexenc_sinkconnect), (gst_speexenc_convert_src),
- (gst_speexenc_convert_sink), (gst_speexenc_get_query_types),
- (gst_speexenc_src_query), (gst_speexenc_init),
- (gst_speexenc_get_tag_value), (comment_init), (comment_add),
- (gst_speexenc_metadata_set1), (gst_speexenc_set_metadata),
- (gst_speexenc_setup), (gst_speexenc_buffer_from_data),
- (gst_speexenc_push_buffer), (gst_speexenc_set_header_on_caps),
- (gst_speexenc_chain), (gst_speexenc_get_property),
- (gst_speexenc_set_property), (gst_speexenc_change_state):
- * ext/speex/gstspeexenc.h:
- Rewrote speex encoder, make sure it can be embedded in ogg.
- Implemented speex decoder.
-
-2004-09-28 Christian Schaller <christian@fluendo.com>
+2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- Remove kioslave plugin. Markey is brewing a new working one
- * ext/Makefile.am: Remove kioslave plugin
- * ext/kio: remove
- * gst-plugins.spec.in: remove kio plugin from spec
-
-2004-09-27 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
- (is_sync_frame), (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_new_client),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Make syncing to keyframes actually work for new clients and lagging
- clients.
+ gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
+ Original commit message from CVS:
+ * gst/playback/Makefile.am:
+ * gst/playback/gstdecodebin.c: (close_pad_link):
+ * gst/playback/gstdecodebin2.c: (analyze_new_pad):
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_handle_message_func), (unknown_type):
+ Let decodebin be the element to post missing-plugin messages for
+ missing decoders (rather than playbin); make playbin implement
+ GstBin::handle_message so we can suppress missing-plugin messages
+ for types we're not handling on purpose (don't want to bring up an
+ installer in those cases).
-2004-09-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/debug/gstnavigationtest.c: (gst_navigationtest_class_init),
- (gst_navigationtest_handle_src_event), (draw_box_planar411),
- (gst_navigationtest_planar411), (gst_navigationtest_change_state):
- * gst/debug/gstnavigationtest.h:
- make navigationtest display button-press and button-release events
+ gst/: Fix potentially unaligned access (#397207).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ * gst-libs/gst/tag/gstvorbistag.c:
+ (gst_tag_list_to_vorbiscomment_buffer):
+ * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
+ Fix potentially unaligned access (#397207).
-2004-09-26 Iain <iaingnome@gmail.com>
+2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (set_scale), (update_scale),
+ (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
+ (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
+ (main):
+ Allow to toggle looping while it plays. Fix callback prototype. Clean
+ up code a bit more. Add copyright header.
-2004-09-26 Iain <iaingnome@gmail.com>
+2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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.
+ sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ Red and blue mask was swapped (spotted by Dan Williams).
-2004-09-25 Christian Schaller <christian@fluendo.com>
+2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * Update SPEC file to be usable in conjunction with Fedora Core,
- Fedora.us and freshrpms packages
- * Fix typo in multifilesrc test Makefile
+ gst-libs/gst/tag/: Use new beats-per-minute tag from core.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Use new beats-per-minute tag from core.
-2004-09-24 Wim Taymans <wim@fluendo.com>
+2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (new_decoded_pad):
- Only signal the no_more_pads signal when we have
- added the stream to our list.
+ po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ Add new files with translatable strings, so they actually make it
+ into the template file one day.
-2004-09-24 Wim Taymans <wim@fluendo.com>
+2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
- * gst/playback/gstplaybasebin.c: (remove_prerolls),
- (new_decoded_pad):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (setup_sinks):
- Don't try to preroll or decode more than one audio/video
- track.
-
-2004-09-24 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
- Throw error if we failed to find a suitable output. This should
- throw an error if we successfully set up a pipeline (e.g. because
- we recognized a media file) but found no decodable streams in it
- (e.g. because it contains only media stream types for which we
- have no decoders, or because it's not a media type).
-
-2004-09-23 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/dirac/Makefile.am:
- * ext/dirac/gstdirac.cc:
- * ext/dirac/gstdiracdec.cc:
- * ext/dirac/gstdiracdec.h:
- Do something. Don't actually know if this works because I don't
- have a demuxer yet.
- * ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
- Add channels=1 to caps returned from _getcaps().
- * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
- (gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
- (gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
- (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
- (gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
- (gst_ogm_parse_change_state):
- Separate between audio/video so ogmaudioparse actually uses the
- audio pad templates. Both audio and video work now, including
- autoplugging. Also use sometimes-srcpad hack.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
- Handle events better. Don't hang on infinite loops.
- * gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
- (gst_avi_demux_init), (gst_avi_demux_reset),
- (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
- (gst_avi_demux_change_state):
- * gst/avi/gstavidemux.h:
- Improve A/V sync. Still not perfect.
- * gst/matroska/ebml-read.c: (gst_ebml_read_seek),
- (gst_ebml_read_skip):
- Handle events better.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (gst_qtdemux_loop_header), (qtdemux_parse_trak),
- (qtdemux_audio_caps):
- Add IMA4. Improve event handling. Save offset after a seek when
- the headers are at the end of the file so that we don't end up in
- an infinite loop.
- * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
- Add low-priority typefind support for files with no length.
-
-2004-09-23 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * testsuite/multifilesink/Makefile.am:
- fix typo
-
-2004-09-22 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls): Fix
- mistakes from thaytan's patches.
-
-2004-09-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
- For completeness, XSync in the destroy function as xvimage does.
-
-2004-09-23 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
- Correct caps negotiation
- * gst/volume/gstvolume.c: (volume_chain_float),
- (volume_chain_int16):
- Modify debug output to be little more informative
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_destroy):
- Add XSync calls after detaching from the shared memory segment to
- avoid a crash.
-
-2004-09-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_loop):
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
- (gst_vorbisenc_chain):
- * ext/vorbis/vorbisenc.h:
- remove explicit newmedia support from oggmux and vorbisenc
- add debug category to vorbisenc
- * gst/multifilesink/gstmultifilesink.c:
- (gst_multifilesink_class_init), (gst_multifilesink_init),
- (gst_multifilesink_dispose), (gst_multifilesink_set_location),
- (gst_multifilesink_set_property), (gst_multifilesink_next_file),
- (gst_multifilesink_handle_event), (gst_multifilesink_chain),
- (plugin_init):
- * gst/multifilesink/gstmultifilesink.h:
- add support for streamheader in multifilesink
-
-2004-09-22 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
- (gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
- (gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
- Prevent infinite loops. More correct error reporting.
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Error out if negotiation fails.
- * gst/playback/gstplaybasebin.c: (setup_source),
- (gst_play_base_bin_change_state), (gst_play_base_bin_error),
- (gst_play_base_bin_found_tag):
- Error/tag forwarding. Pre-roll fixes for source errors on state
- changes (e.g. "file does not exist") to prevent hangs.
-
-2004-09-21 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * testsuite/multifilesink/Makefile.am:
- * testsuite/multifilesink/lame_test.c: (gst_newmedia_base_init),
- (gst_newmedia_class_init), (gst_newmedia_init),
- (gst_newmedia_chain), (gst_newmedia_trigger), (test_format),
- (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/multifilesrc_test.c: (main):
- * testsuite/multifilesink/oggtheora_test.c:
- (gst_newmedia_base_init), (gst_newmedia_class_init),
- (gst_newmedia_init), (gst_newmedia_chain), (gst_newmedia_trigger),
- (test_format), (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/oggvorbis_test.c:
- (gst_newmedia_base_init), (gst_newmedia_class_init),
- (gst_newmedia_init), (gst_newmedia_chain), (gst_newmedia_trigger),
- (test_format), (newfile_signal), (test_signal), (main):
- * testsuite/multifilesink/wavenc_test.c: (gst_newmedia_base_init),
- (gst_newmedia_class_init), (gst_newmedia_init),
- (gst_newmedia_chain), (gst_newmedia_trigger), (test_format),
- (newfile_signal), (test_signal), (main):
- New media tests
-
-2004-09-20 Christian Schaller <christian@fluendo.com>
-
- * Fix mikmod license to LGPL as they have relicensed
- * Move Dirac and Effectv into LGPL section of README_license
-
-2004-09-20 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mad/gstmad.c: (gst_mad_check_caps_reset),
- (gst_mad_change_state):
- Allow for mp3 rate/channels changes. However, only very
- conservatively. Reason that we *have* to enable this is smiply
- because the mad find_sync() function is not good enough, it will
- regularly sync on random data as valid frames and therefore make
- us provide random caps as *final* caps of the stream. The best fix
- I could think of is to simply require several of the same stream
- changes in a row before we change caps.
- The actual testcase that works now is #
- * ext/ogg/Makefile.am:
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstogmparse.c:
- OGM support (video only for now; I need an audio sample file).
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
- (gst_asf_demux_add_video_stream):
- WMV extradata.
- * gst/playback/gstplaybasebin.c: (unknown_type):
- Don't error out on single unknown-types after all. It's wrong.
- If we found type of video and audio but not of a subtitle stream,
- it will still error out (which is unwanted). Will find a better fix
- later on.
- * gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
- (ogmaudio_type_find), (plugin_init):
- OGM support.
-
-2004-09-20 Johan Dahlin <johan@gnome.org>
-
- * ext/jpeg/gstjpegdec.c (gst_jpegdec_chain): Allocate the buffer
- after setting caps.
-
-2004-09-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
-
- * gst/wavenc/gstwavenc.c: (gst_wavenc_init), (gst_wavenc_chain):
- * gst/wavenc/gstwavenc.h:
- Added newmedia support to wavenc
-
-2004-09-17 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gstfdset.c: (gst_fdset_fd_has_closed),
- (gst_fdset_fd_has_error), (gst_fdset_fd_can_read),
- (gst_fdset_fd_can_write), (gst_fdset_wait):
- * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_remove), (gst_multifdsink_get_stats),
- (gst_multifdsink_remove_client_link),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_handle_clients),
- (gst_multifdsink_close), (gst_multifdsink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
- (gst_tcpserversink_removed):
- Small cleanups in fdset.c
- Use a hastable to map fd to the client structure for faster
- lookup in _remove and get_stats.
- Added virtual function to close the fds.
- Handle clients even when the select/poll call was unblocked because
- of a command.
- Implement syncing to keyframe in the recovery procedure.
+ * 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.
-2004-09-16 Iain <iaingnome@gmail.com>
+2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int): Free the
- try caps.
+ * gst-plugins-base.spec.in:
+ add latest files
+ Original commit message from CVS:
+ add latest files
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.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.
+ gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/mixerutils.h:
+ Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
+ used when compiling with c++ compilers as well.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.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.
+ gst/typefind/gsttypefindfunctions.c: Fix comment.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c:
+ Fix comment.
-2004-09-15 Arwed v. Merkatz <v.merkatz@gmx.net>
+2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.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.
+ gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (post_missing_element_message),
+ (gen_video_element), (gen_text_element), (gen_audio_element),
+ (gen_vis_element):
+ Post missing-plugin messages also when we error out because
+ converters, textoverlay or auto*sinks are missing (#161922).
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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.
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_stream_data):
- Try to fix a/v sync issues.
+2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-15 David Schleef <ds@schleef.org>
+ 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.
- * configure.ac: remove NASM check, since we don't use it. Update
- dirac check to 0.4
- * ext/dirac/gstdiracdec.cc: update to current 0.4 API
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Initialized variables.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
- (gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
- SVQ3 format
+2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * 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?
+2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Added docs.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
+ (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
+ (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
+ (gst_ogg_demux_perform_seek):
+ * ext/ogg/gstoggdemux.h:
+ Added docs.
+ Add some more comments.
+ Small cleanups.
+
+2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ Small documentation updates/fixes
+ Original commit message from CVS:
+ * ext/theora/theoradec.c:
+ * ext/vorbis/vorbisdec.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Small documentation updates/fixes
+
+2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
+ Original commit message from CVS:
+ * configure.ac:
+ Require core CVS HEAD for Andy's basesrc/sink API additions.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
- * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
- Volume is a double not a float.
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
+ Original commit message from CVS:
+ Patch by: Günter Thelen <daedalus dot inc at gmx net>
+ * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
+ (plugin_init):
+ Add typefinder for flac-in-ogg in conformance with the ogg-mapping
+ on flac.sf.net (there appear to be other versions of the first
+ ogg page in the wild) (#391365).
-2004-09-15 Wim Taymans <wim@fluendo.com>
+2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_remove_client_link),
- (gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
- Don't close the fd in multifdsink as we didn't open it in the
- first place. Some cleanups.
+ configure.ac: Check if localtime_r() is available.
+ Original commit message from CVS:
+ * configure.ac:
+ Check if localtime_r() is available.
+ * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
+ If localtime_r() is not available, fall back to localtime(). Should
+ fix build on MingW (#393310).
-2004-09-15 Wim Taymans <wim@fluendo.com>
+2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- Fix the case where the muxer would mark pages as delta
- frames when they are not (vorbis only ogg).
+ gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (parse_mdvdsub):
+ * gst/subparse/gstsubparse.h:
+ Remove spurious 1000 subtrahend when calculating the timestamp from
+ the frame number and the frame rate . Also, use the frames/second
+ value specified in the first line of the file, if one is specified
+ there. Should fix #357503.
+ * tests/check/elements/subparse.c: (do_test),
+ (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
+ (subparse_suite):
+ Add some basic unit tests for the microdvd subtitle format.
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
+
+ sys/xvimage/xvimagesink.c: Fixes : #390076.
+ Original commit message from CVS:
+ 2007-01-07 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_xvimage_put),
+ (gst_lookup_xv_port_from_adaptor),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_set_xwindow_id),
+ (gst_xvimagesink_set_event_handling),
+ (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
+ (gst_xvimagesink_init), (gst_xvimagesink_class_init):
+ Patch by : Young-Ho Cha <ganadist at chollian dot net>
+ Fixes : #390076.
+ Add an adaptor property to select a specific XV adaptor.
+ * sys/xvimage/xvimagesink.h:
+
+2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Use flow_lock much more to protect every access to xwindow.
+ Original commit message from CVS:
+ 2007-01-07 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
+ (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
+ (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
+ (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
+ (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
+ (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
+ (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_change_state),
+ (gst_xvimagesink_set_xwindow_id),
+ (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
+ Use flow_lock much more to protect every access to xwindow.
+ Try to catch erros while creating images in case some drivers
+ are
+ just generating an XError when the requested image is too big.
+ Should fix : #354698, #384008, #384060.
+ * tests/icles/stress-xoverlay.c: (cycle_window),
+ (create_window):
+ Implement some stress testing of setting window xid.
+
+2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/common/libgsaudio.def: Add new exported function.
+ Original commit message from CVS:
+ * win32/common/libgsaudio.def:
+ Add new exported function.
+ * win32/common/libgstogg.dsp:
+ Add gstoggaviparse.c to the build.
+ * win32/common/libgstvideoscale.dsp:
+ Add vs_4tap.c to the build.
+ * win32/common/libgstvorbis.dsp:
+ Add vorbistag.c to the build.
- * gst/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).
+2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
-2004-09-15 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * 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>
- * 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.
+ 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.
-2004-09-14 Brian Cameron <brian.cameron@sun.com
+2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.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.
+ ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
+ (gst_ogm_parse_change_state):
+ Activate pads before adding them to the element.
-2004-09-17 Wim Taymans <wim@fluendo.com>
+2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/playback/gstplaybasebin.c: (queue_overrun), (no_more_pads),
- (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_add_element):
- * gst/playback/gstplaybin.c: (gst_play_bin_send_event):
- Some more work on making sure seeking pauses the pipeline and
- that changing the uri actually does something.
+ tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
+ Original commit message from CVS:
+ * tests/examples/seek/scrubby.c: (main):
+ * tests/examples/seek/seek.c: (main):
+ Call g_thread_init() first thing in main() (see #391278).
-2004-09-17 Wim Taymans <wim@fluendo.com>
+2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstfdset.c: (gst_fdset_wait):
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_close):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send),
- (gst_tcpserversink_close):
- Be a bit more paranoid when freeing memory.
+ tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/libs/.cvsignore:
+ * tests/check/libs/netbuffer.c: (GST_START_TEST),
+ (netbuffer_suite):
+ Add test for GstNetBuffer + gst_buffer_copy(). Disabled
+ for the time being, since it's broken, see #393099.
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream),
- (qtdemux_parse_trak):
- Don't crash by dividing by zero (see sample movie in #126922).
+ tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Update to use GST_PLUGINS_BASE_CFLAGS as well.
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Don't touch non-existing data (fixes crash on file in #140147).
+ 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
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
- * gst/playback/gstplaybasebin.c:
- (gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
- Handle double disposals, and proper change of URIs.
+ 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).
-2004-09-13 Martin Eikermann <meiker@upb.de>
+2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegparse.c:
- fix synchronistation for streams recorded from digital PCR
- fixes bug #119376
+ Dist design docs.
+ Original commit message from CVS:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/design/Makefile.am:
+ Dist design docs.
-2004-09-13 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+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
- * 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>
+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)
- * configure.ac:
- bump nano to cvs
+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
-=== release 0.8.4 ===
+2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac: releasing 0.8.4, "Alias"
+2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-31 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * ext/theora/Makefile.am:
- fix makefile. Fixes #151462.
+2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-30 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- Patch by: David Schleef
+2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * sys/Makefile.am:
- rename our detection macro for V4L2. Fixes #151236.
+ 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.
-2004-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- Patch by: David Schleef
+ add doap file
+ Original commit message from CVS:
+ * Makefile.am:
+ * gst-plugins-base.doap:
+ * gst-plugins-base.spec.in:
+ add doap file
- * configure.ac:
- check to define LAMEPRESET. Fixes #151232.
+2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
-2004-08-27 David Schleef <ds@schleef.org>
+ Declare variables at the beginning of a block. Fixes #383195.
+ Original commit message from CVS:
+ Patch by: Jens Granseuer <jensgr at gmx net>
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ (gst_base_rtp_audio_payload_handle_frame_based_buffer),
+ (gst_base_rtp_audio_payload_handle_sample_based_buffer):
+ * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
+ Declare variables at the beginning of a block. Fixes #383195.
- * 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.
+2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ configure.ac: Bump version nano - back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump version nano - back to CVS.
- * configure.ac:
- bump nano for prerelease
+=== release 0.10.11 ===
-2004-08-27 David Schleef <ds@schleef.org>
+2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * sys/sunaudio/Makefile.am: Add sunaudiosrc patch from Bala
- * sys/sunaudio/gstsunaudiosrc.c:
- * sys/sunaudio/gstsunaudiosrc.h:
+ configure.ac: releasing 0.10.11, "Dumb things"
+ Original commit message from CVS:
+ === release 0.10.11 ===
+ 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
+ * configure.ac:
+ releasing 0.10.11, "Dumb things"
-2004-08-27 Arwed v. Merkatz <v.merkatz@gmx.net>
+2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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
+ gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
+ (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
+ Handle the case where an element has multiple pads with
+ unfixed caps as well as still possibly producing more dynamic
+ pads by storing each case as a distinct entry in the dynamic list.
+ Fixes #38223 again.
-2004-08-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
- patch by: Zaheer Abbas Merali
+ 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/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>
+2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
- * configure.ac:
- Clean up the test for lame presets
+ * po/.gitignore:
+ Ignore all pot files
+ Original commit message from CVS:
+ Ignore all pot files
-2004-08-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.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.
+ gst/audiorate/gstaudiorate.c: Delete bad debug code.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Delete bad debug code.
+ Fixes #381219
-2004-08-18 Wim Taymans <wim@fluendo.com>
+2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.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.
+ Fix compilation on win32 under VS8
+ Original commit message from CVS:
+ * gst/videoscale/vs_4tap.c:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ * win32/vs8/libgstvideoscale.vcproj:
+ Fix compilation on win32 under VS8
+ Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
+ Partially fixes #381175
-2004-08-18 Wim Taymans <wim@fluendo.com>
+2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.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.
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-08-17 David Schleef <ds@schleef.org>
+2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: Fix caps to handle full range
- of rates and channels. Make debugging less obnoxious.
+ 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.
- 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:
+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
-2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioscale/gstaudioscale.c:
- * gst/audioscale/gstaudioscale.h:
- made audioscale resample from any sample rate to any sample rate
+ 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.
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/libpng/gstpngdec.c:
- error out on unsupported types
+ 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.
-2004-08-17 Iain <iaingnome@gmail.com>
+2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
- * ext/flac/gstflacenc.c (gst_flacenc_update_quality): Only set the
- mid_side and loose_mid_side properties if its a stereo stream.
+ 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.
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_iterate),
- (gst_ogg_pad_push):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_request_new_pad), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_push_page),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- Mark delta units in the muxer.
- Try to decode the packet after an out-of-sync error from
- libogg.
+ 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.
-2004-08-17 Wim Taymans <wim@fluendo.com>
+2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_set_property), (gst_multifdsink_get_property):
- * gst/tcp/gstmultifdsink.h:
- Added option to send a keyframe to clients as the first buffer.
- Make timeout property writable.
+ gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Avoid integer underflow when the found probability for mp3 is
+ smaller than the 'penalty' we subtract if there's not a clean
+ mp3 header sync at offset 0.
-2004-08-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- patch by: Wim Taymans
+ 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
- * gst/tcp/gstfdset.c:
- * gst/tcp/gstmultifdsink.c:
- fix index comparison, should include 0
+2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-16 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/tcp/gstfdset.c: (ensure_size), (gst_fdset_new),
- (gst_fdset_add_fd), (gst_fdset_remove_fd),
- (gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
- (gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
- (gst_fdset_wait):
- copy when reallocing for poll so the select arguments don't get
- changed during the call
+2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-08-16 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/theora/theoraenc.c: (gst_border_mode_get_type),
- (gst_theora_enc_class_init), (theora_enc_sink_link),
- (theora_buffer_from_packet), (theora_enc_chain):
- Fix bug where buffers were not marked as keyframes
- correctly.
+2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-15 Zaheer Abbas Merali <zaheerabbas at merali 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
- * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type),
- (gst_lame_preset_get_type), (gst_lame_class_init):
- describe the enum values for vbr mode and presets more verbosely
+2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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).
- * 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
+2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * ext/lame/gstlame.c: (gst_lame_get_property):
- whoops forgot break, thanks teuf
+2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
-2004-08-13 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
+ (gst_audio_rate_sink_event), (gst_audio_rate_chain):
+ Fix audiorate, so that it accurately sets offsets and timestamps.
+ Doesn't change the fundamental algorithmic decisions; so should be
+ safe.
+ * tests/check/Makefile.am:
+ Enable audiorate test now that it passes.
- * 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
+2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
+ clear xv when going to NULL, remove // commented non-existant proto
+ * tests/examples/seek/seek.c: (main):
+ add missing tooltip description for scrub and play_scrub
- * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_init):
- added getcaps function so samplerate doesnt get fixated to silly values
+2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ configure.ac: Bump liboil requirement to 0.3.8.
+ Original commit message from CVS:
+ * configure.ac:
+ Bump liboil requirement to 0.3.8.
+ * gst-libs/gst/riff/riff-media.c:
+ Add Dirac fourcc.
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.h:
+ Use liboil's stdint.h.
+ * gst/videotestsrc/videotestsrc.c:
+ Remove liboil related ifdef's, since they aren't needed now, and
+ won't work with future versions.
- * ext/lame/gstlame.c: (gst_lame_src_link):
- revert previous fix
+2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
+
+ gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
+ Original commit message from CVS:
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ * gst/videoscale/vs_4tap.c:
+ * gst/videoscale/vs_4tap.h:
+ * gst/videoscale/vs_image.c:
+ * gst/videoscale/vs_image.h:
+ * gst/videoscale/vs_scanline.c:
+ * gst/videoscale/vs_scanline.h:
+ Add a 4-tap image scaler. Theoretically looks much prettier.
+ The tap calculation could use some improvement.
-2004-08-12 Johan Dahlin <johan@gnome.org>
+2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
+
+ Various gsize and gssize printf fixes. Fixes #372507.
+ Original commit message from CVS:
+ Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
+ (gst_riff_parse_strf_iavs):
+ * gst/subparse/gstsubparse.c: (convert_encoding):
+ * gst/tcp/gstmultifdsink.c:
+ (gst_multi_fd_sink_handle_client_write):
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
+ (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
+ (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
+ * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
+ (gst_ximagesink_ximage_new):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
+ Various gsize and gssize printf fixes. Fixes #372507.
+
+2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
+ (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
+ (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
+ (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
+ (vorbis_dec_chain_forward), (vorbis_dec_chain):
+ * ext/vorbis/vorbisdec.h:
+ First stab at vorbis reverse playback.
+
+2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_event), (gst_base_audio_sink_render):
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ Make the clock sync code more accurate wrt resampling and playback
+ at different rates.
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Use better algorithm to interpolate sample rates.
+
+2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
+
+ ext/ogg/gstoggdemux.c: Improve a debug line slightly.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
+ Improve a debug line slightly.
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
+ Call gst_riff_init() in plugin_init, to avoid getting errors from
+ the debug system (unrelated changes to another plugin made this turn
+ up; not sure why).
- * 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.
+2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ win32/common/libgsttag.def: Add missing symbol (#366492).
+ Original commit message from CVS:
+ Patch by: Sergey Scobich <sergery.scobich at gmail com>
+ * win32/common/libgsttag.def:
+ Add missing symbol (#366492).
- * 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
+2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-12 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst/playback/gststreamselector.c: Don't unref a NULL pad.
+ Original commit message from CVS:
+ * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
+ Don't unref a NULL pad.
- * 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
+2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
-2004-08-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init):
- set default compression ratio parameter to 0.0 so bitrate parameter
- works :)
+2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-08-11 David Schleef <ds@schleef.org>
+ 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
- * gst/tcp/gstfdset.c: Fix compile problem on OS/X.
+2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-11 David Schleef <ds@schleef.org>
+ 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/mpeg1sys/gstmpeg1systemencode.c: Oops, this was correct
- before.
+2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
-2004-08-11 David Schleef <ds@schleef.org>
+ 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-libs/gst/video/videosink.h: Change copyright block to LGPL.
+2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
-2004-08-11 David Schleef <ds@schleef.org>
+ gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
+ Recognise more mpeg4 elementary video streams.
- * ext/pango/gsttextoverlay.c: Add copyright block and fix plugin
- license field
- * gst-libs/gst/idct/Makefile.am: Remove mmx/sse code
- * gst-libs/gst/video/gstvideosink.c: Change copyright block to
- LGPL.
- * gst/auparse/gstauparse.c: Fix plugin license field.
- * gst/monoscope/gstmonoscope.c: Fix plugin license field.
- * gst/mpeg1sys/gstmpeg1systemencode.c: Fix plugin license field.
- * gst/rtp/gstrtp.c: Fix plugin license field.
+2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Lower the probability of mp3 typefinding functions if we don't find a
+ valid mp3 header at the start of the file.
+ Closes #369482
+
+2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (theora_dec_sink_event), (theora_dec_chain_forward),
+ (theora_dec_flush_decode), (theora_dec_chain_reverse),
+ (theora_dec_chain):
+ Document and partially implement an algorithm for doing reverse playback
+ of theora video.
+
+2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
+
+ win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
+ Original commit message from CVS:
+ Patch by: Sergey Scobich <sergey.scobich at gmail com>
+ * win32/common/config.h:
+ * win32/common/interfaces-enumtypes.c:
+ * win32/common/libgsttag.def:
+ * win32/vs8/gst-plugins-base.sln:
+ * win32/vs8/libgstaudioresample.vcproj:
+ * win32/vs8/libgstinterfaces.vcproj:
+ * win32/vs8/libgstogg.vcproj:
+ * win32/vs8/libgstriff.vcproj:
+ * win32/vs8/libgsttag.vcproj:
+ * win32/vs8/libgsttheora.vcproj:
+ * win32/vs8/libgstvideoscale.vcproj:
+ * win32/vs8/libgstvorbis.vcproj:
+ Misc. VS8 build fixes: fix syntax in config.h, add missing entries
+ to libgsttag.def; add missing dependencies for some vs8 projects;
+ re-arrange placement of .def files in vs8 projects (#366334).
-2004-08-11 Wim Taymans <wim@fluendo.com>
+2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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).
+ ext/ogg/gstogg.c: Remove unused variable.
+ Original commit message from CVS:
+ * ext/ogg/gstogg.c:
+ Remove unused variable.
+ * ext/ogg/gstoggdemux.c:
+ Fix Wim's surname in plugin description.
-2004-08-11 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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>
+ 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.
- * po/LINGUAS:
- * po/sq.po:
- Added Albanian translation (Laurent Dhima)
- * po/cs.po:
- updated
+2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * ext/lame/gstlame.c:
- fix/add debugging
+2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * sys/ximage/ximagesink.c:
- * sys/xvimage/xvimagesink.c:
- assign all TOO_LAZY's to a real category. Thanks to Warthy Warthog.
+2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-06 Wim Taymans <wim@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/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.
+2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/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
+2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-08-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * po/nl.po:
- * po/sv.po:
- updated translations
+2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-08-04 Benjamin Otte <otte@gnome.org>
+ * ChangeLog:
+ I'm too lazy to comment this
+ Original commit message from CVS:
+ *** empty log message ***
- * 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>
+2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
- * 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
+ ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
+ Original commit message from CVS:
+ Patch by: Mark Nauwelaerts <manauw at skynet dot be>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
+ Remove the pad from the element in release_pad.
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/uk.po:
- updated translation
+ sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
+ Explicitly create our custom buffer classes at a thread-safe
+ location as well, since g_type_class_ref() doesn't seem to be
+ entirely thread-safe either (#365501; also see #349410).
-2004-08-03 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
- add debugging for display PAR calculation
+ gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
+ (gst_riff_parse_info):
+ If strings in INFO chunk are not UTF-8, do something similar to
+ what we do for ID3v1 tags: check a number of environment variables
+ (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
+ character sets to try, otherwise try the current locale and/or fall
+ back on ISO-8859-1. Fixes #360552.
-2004-08-02 David Schleef <ds@schleef.org>
+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.
- * configure.ac: Fix mikmod CFLAGS.
+2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
+ Original commit message from CVS:
+ * gst/subparse/Makefile.am:
+ * gst/subparse/gstsubparse.c:
+ (gst_sub_parse_data_format_autodetect),
+ (gst_sub_parse_format_autodetect), (handle_buffer),
+ (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
+ * gst/subparse/gstsubparse.h:
+ * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
+ (parse_tmplayer):
+ * gst/subparse/tmplayerparse.h:
+ Add support for TMPlayer-type subtitles (#362845).
+ * tests/check/elements/subparse.c: (test_tmplayer_do_test),
+ (GST_START_TEST), (subparse_suite):
+ Add some basic unit tests for the above.
-2004-07-27 Benjamin Otte <otte@gnome.org>
+2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
+ Original commit message from CVS:
+ * tests/check/elements/audiorate.c: (test_injector_base_init),
+ (test_injector_class_init), (test_injector_chain),
+ (test_injector_init), (probe_cb), (do_perfect_stream_test),
+ (GST_START_TEST), (audiorate_suite):
+ More tests for audiorate: inject buffers to check behaviour when
+ buffers overlap.
-2004-08-02 Iain <iain@prettypeople.org>
+2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
+ (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
+ Add some basic unit tests for audiorate. Disabled at the moment
+ since it doesn't pass yet (see bug #363119).
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: bump nano back to development
+ 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.
-=== release 0.8.3 ===
+2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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)
- * configure.ac: releasing 0.8.3, "Water"
+2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
-2004-08-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- patch from: Benjamin Otte
+2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lame/gstlame.c: (gst_lame_src_link), (gst_lame_init):
- add link function to lame. Fixes #148986.
+ 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).
-2004-08-02 Johan Dahlin <johan@gnome.org>
+2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/multipart/multipartmux.c (gst_multipart_mux_next_buffer):
- fix debugging log
+ 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.
-2004-07-30 David Schleef <ds@schleef.org>
+2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videomixer/Makefile.am: Fix things that should have been
- fixed in the last checkin.
+ 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.
-2004-07-30 David Schleef <ds@schleef.org>
+2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/multipart/Makefile.am: Fix things that should have been
- fixed in the last checkin.
+ 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.
-2004-07-30 David Schleef <ds@schleef.org>
+2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
- * testsuite/multifilesink/Makefile.am: Fix unused variable.
+ 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.
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * 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
+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.
-2004-07-30 Wim Taymans <wim@fluendo.com>
+2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_remove),
- (gst_multifdsink_clear), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
- * gst/tcp/gstmultifdsink.h:
- Recover from a select with a bad file descriptor by removing
- the client.
+ gst/playback/gstplaybin.c: The old pad activation spiel.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (setup_sinks):
+ The old pad activation spiel.
-2004-07-30 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- fix requirement of core
- * gst-libs/gst/play/play.c: (gst_play_error_plugin),
- (gst_play_pipeline_setup):
- don't use colorspace element. do use hermescolorspace element.
- make macro to get a colorspace element.
- mark strings for translation.
- * po/POTFILES.in:
- add play.c
- * po/af.po:
- * po/az.po:
- * po/cs.po:
- * po/en_GB.po:
- * po/hu.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- * po/uk.po:
- update translations
+ gst/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).
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
- fix default for newmedia flag
+ 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.
-2004-07-30 Wim Taymans <wim@fluendo.com>
+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.
- * 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.
+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).
-2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
+
+ tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
+ Original commit message from CVS:
+ 2006-10-10 Julien MOUTTE <julien@moutte.net>
+ * tests/examples/seek/seek.c: (do_seek),
+ (rate_spinbutton_changed_cb): When changing spinbutton we try
+ to change the rate on the fly.
+
+2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/: Add WMS caps.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Add WMS caps.
+
+2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
+
+ ext/gnomevfs/: Fix URI interface implementation return type.
+ Original commit message from CVS:
+ 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ Patch by: Josep Torre Valles <josep@fluendo.com>
+ * ext/gnomevfs/gstgnomevfssink.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Fix URI interface implementation return type.
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
+ Fix what looks like a copy/paste issue when assigning values.
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_get_type):
+ Cast to prevent Forte warnings.
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ Fix URI interface implementation return type.
+ gst_pad_query_position requires a signed integer pointer as
+ 3rd parameter, GstClockTime is unsigned.
+ * gst/audioconvert/audioconvert.c:
+ Fix integer overflow when treated as signed.
+ * gst/audioresample/resample.c: (resample_add_input_data):
+ Cast to prevent warnings on Forte.
+ * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
+ Fix integer overflow when treated as signed.
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
+ * gst/playback/gstdecodebin.c: (queue_filled_cb),
+ (cleanup_decodebin):
+ Who initialises a guint to -1!
+ Cast function pointers to prevent warnings on Forte.
+ * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
+ (queue_threshold_reached):
+ Cast function pointers correctly to prevent warnings on Forte.
+ * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
+ Cast function pointers correctly to prevent warnings on Forte.
+ * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
+ Obvious change to unsigned, 0xEF > max signed char.
+ * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
+ GstClockTime is unsigned, initialise correctly.
+ * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
+ Cast so pointer arithemetic doesn't cause warnings on Forte.
+ * gst/videorate/gstvideorate.c:
+ Use correct return value.
+ * tests/examples/seek/scrubby.c:
+ GstClockTime is unsigned, initialise correctly.
+
+2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
+
+ gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
+ Original commit message from CVS:
+ Patch by: Ferenc Gerlits <fgerlits at gmail com>
+ * gst/typefind/gsttypefindfunctions.c:
+ Recognise XML files and XML-like files shorter than 256 bytes as
+ well (fixes #359237).
- * 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
+2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
-2004-07-30 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.
- * 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.
+2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-29 David Schleef <ds@schleef.org>
+ gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
+ Some more guards against invalid input.
- * configure.ac: make test for audiofile more strict
+2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
-2004-07-25 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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/typefind/gsttypefindfunctions.c: (plugin_init):
- give different names to typefind functions
+2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-28 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ Unref message in doc-example (spotted by Robert McQueen)
- * 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
+2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-27 Steve Lhomme <steve.lhomme@free.fr>
+ 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.
- * win32/gst.sln:
- * gst-libs/gst/*/*.vcproj:
- * gst/*/*.vcproj:
- more working plugins
+2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-27 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * testsuite/alsa/Makefile.am:
- * testsuite/alsa/srcstate.c:
- add test for alsasrc changing state
+2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
-2004-07-27 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.
- * gst/silence/gstsilence.c: (gst_silence_init), (gst_silence_link),
- (gst_silence_get):
- * gst/silence/gstsilence.h:
- fix silence generation for 16bit raw audio
+2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_change_state):
+ Also call parent state change function to activate pads.
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
+ (mpeg1_parse_header), (mpeg1_sys_type_find):
+ Add some more debug info in mpeg typefinding.
- * 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.
+2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
-2004-07-27 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_chain):
+ Zero byte theora packets are valid and well-defined; don't warn on
+ them.
- * 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.
+2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-26 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_get_stats), (find_limits),
+ (gst_multi_fd_sink_queue_buffer):
+ API: add dropped_buffers to the get-stats GValueArray
- * gst/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>
+2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/audio/gstaudiofilter.c:
- Clean the local include
+ 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.
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
+2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-2004-07-25 Steve Lhomme <steve.lhomme@free.fr>
+2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/matroska/ebml-write.c:
- Fix some 64 bits constants to be glib friendly
+ * 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.
-2004-07-24 Steve Lhomme <steve.lhomme@free.fr>
+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.
- * win32/gst.sln:
- * gst-libs/gst/gst-libs.def:
- * gst-libs/gst/gst-libs.vcproj:
- * gst/matroska/matroska.def:
- * gst/matroska/matroska.vcproj:
- Add the preliminary canvas to build plugins on Win32
+2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-23 Benjamin Otte <otte@gnome.org>
+ 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.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- don't enfore negotiation from source side, it breaks
- sinesrc ! audioconvert ! osssink
+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.
-2004-07-22 David Schleef <ds@schleef.org>
+2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add typefind
- for ELF files, since they can easily be recognized as audio/mpeg.
- (bug #147441)
+ 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.
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ tests/check/pipelines/oggmux.c: Remove obsolete comment.
+ Original commit message from CVS:
+ * tests/check/pipelines/oggmux.c:
+ Remove obsolete comment.
-2004-07-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
- * 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.
+ 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.
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+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?)
- * gst/videoscale/videoscale.c: (gst_videoscale_planar411),
- (gst_videoscale_scale_nearest_16bit):
- comment algorithm
+2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-07-22 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_class_init), (gst_videotestsrc_src_link),
- (gst_videotestsrc_init), (gst_videotestsrc_get),
- (gst_videotestsrc_set_pattern), (gst_videotestsrc_set_property),
- (gst_videotestsrc_get_property):
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- cleanup and commenting
-
-2004-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
- (gst_ogg_demux_get_formats), (gst_ogg_demux_src_query),
- (gst_ogg_demux_src_event), (gst_ogg_demux_src_convert),
- (gst_ogg_demux_handle_event), (gst_ogg_demux_seek_before),
- (_find_chain_get_unknown_part), (_find_streams_check),
- (gst_ogg_demux_push), (gst_ogg_pad_push):
- * ext/theora/theoradec.c: (theora_get_formats),
- (theora_dec_src_convert), (theora_dec_sink_convert),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (vorbis_dec_convert), (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_event):
- More seeking fixes, oggdemux now supports seeking to time and
- uses the downstream element to convert granulepos to time.
- Seeking in theora-only ogg files now works.
-
-2004-07-21 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (gst_theora_dec_init),
- (theora_get_formats), (theora_get_event_masks),
- (theora_get_query_types), (theora_dec_src_convert),
- (theora_dec_sink_convert), (theora_dec_src_query),
- (theora_dec_src_event), (theora_dec_event), (theora_dec_chain):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (vorbis_get_event_masks), (vorbis_get_query_types),
- (gst_vorbis_dec_init), (vorbis_dec_convert),
- (vorbis_dec_src_query), (vorbis_dec_src_event), (vorbis_dec_event):
- Added query/convert/formats functions to vorbis and theora decoders
- so that the outside world can use them too. Fixed seeking on an
- ogg/theora/vorbis file by disabling the seeking seeking on the
- theora srcpad.
-
-2004-07-21 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_renegotiate_size), (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_set_xwindow_id): Optimize
- images creation for both elements. We don't create the image on caps
- nego or renego, we just destroy the internal one if present if it does
- not match the needs. The chain function takes care of creating a new
- image when needed.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_xwindow_decorate), (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_xwindow_id): Additionally xvimage now contains
- the image format information. The buffer pool checks for the context
- image format and discard images with different formats.
- * sys/xvimage/xvimagesink.h: Adding im_format in the xvimage structure.
-
-2004-07-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- no point in doing any chaining if the pad we want to push from
- isn't usable.
+ gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (remove_sinks):
+ Only remove visualisation from visbin if there is a visbin (or:
+ don't throw warnings when closing totem without playing a file).
-2004-07-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+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-libs/gst/riff/riff-media.c:
- (gst_riff_create_audio_caps_with_data):
- Fix double end-to-native symbol conversion (#148021).
+ * gst-plugins-base.spec.in:
+ add new header file to spec
+ Original commit message from CVS:
+ add new header file to spec
-2004-07-20 David Schleef <ds@schleef.org>
+2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_decorate):
- Don't use an Atom that doesn't exist.
+ tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Another attempt to make the gen64 buildbot happy.
-2004-07-20 Wim Taymans <wim@fluendo.com>
+2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
- * 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.
+ ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
+ Original commit message from CVS:
+ Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
+ * ext/libvisual/visual.c: (gst_visual_clear_actors),
+ (gst_visual_chain), (gst_visual_change_state):
+ Libvisual plugin was not passing audio data to libvisual 0.4.0
+ correctly. Fixes #357800
-2004-07-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * po/LINGUAS:
- * po/hu.po:
- added Hungarian translation (Laszlo Dvornik)
+ tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
+ Original commit message from CVS:
+ * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
+ Add timeout to _get_state() so we see which pipeline it is
+ that causes trouble on the gen64 build bot.
-2004-07-20 Wim Taymans <wim@fluendo.com>
+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.
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer):
- * gst/tcp/gsttcp-marshal.list:
- Fixed the stupid marshal definition.
+2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
+ Original commit message from CVS:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ Moved AudioCodecType into priv
+ Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
+
+2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
+ (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
+ (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
+ (new_pad):
+ Cleanups and small leak fixes.
+ Added Depayloaders to valid list of autopluggable elements.
+
+2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
+ (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
+ (gen_video_element), (gen_text_element), (gen_audio_element),
+ (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
+ (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
+ Detect NO_PREROLL state change returns and disable clock distribution to
+ the sinks so that sync is disabled.
+ Avoid some type checking and do simple casts instead.
+ Small cleanups, fix some FIXMEs.
+ Be more robust when linking user specified elements, catch an report
+ errors. Fixes #357404.
+ Fix some leaks in the error paths.
+
+2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ ChangeLog surgery for missing bug-number
+ Original commit message from CVS:
+ ChangeLog surgery for missing bug-number
- * gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_init), (gst_multifdsink_add),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_queue_buffer), (gst_multifdsink_chain),
- (gst_multifdsink_set_property), (gst_multifdsink_get_property),
- (gst_multifdsink_init_send):
- * gst/tcp/gstmultifdsink.h:
- Added more stats, added timeout for a client, fixed some typos
- and added some comments.
+2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
-2004-07-20 Wim Taymans <wim@fluendo.com>
+ 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/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
- (gst_multifdsink_add), (gst_multifdsink_get_stats),
- (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_write):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcp-marshal.list:
- Added get_stats method that returns a GValueArray of
- stats values.
+2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-19 Benjamin Otte <otte@gnome.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.
- * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
- make sure longname, description and author are valid UTF-8
+2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/ximage/ximagesink.c: (gst_ximagesink_change_state),
- (gst_ximagesink_set_property):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_property):
- make sure SYNCHRONOUS is respected after getting the X context
+2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ tests/check/Makefile.am: See if this makes the build bots happy.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ See if this makes the build bots happy.
+ * tests/check/libs/cddabasesrc.c:
+ UTF8-ise my name.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_handle_src_event),
- (gst_matroska_demux_parse_blockgroup):
- * gst/matroska/matroska-ids.h:
- add BlockReference tag and ignore it to clear out log.
- ignore NAVIGATION events to clear out log.
+2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-07-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst/subparse/samiparse.c: (handle_start_font),
+ (fix_invalid_entities):
+ More case-insensitivity for certain tags; recognise entities with
+ decimal codes as special entities as well (#357330).
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
- (gst_matroska_demux_add_stream):
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init):
- add debug categories
+2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-16 Wim Taymans <wim@fluendo.com>
+ gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
+ Original commit message from CVS:
+ * gst-libs/gst/Makefile.am:
+ Need to build tag directory before cdda.
- * ext/libpng/Makefile.am:
- * ext/libpng/gstpng.c: (plugin_init):
- * ext/libpng/gstpngdec.c: (user_error_fn), (user_warning_fn),
- (gst_pngdec_get_type), (gst_pngdec_base_init),
- (gst_pngdec_class_init), (gst_pngdec_sinklink), (gst_pngdec_init),
- (gst_pngdec_src_getcaps), (user_read_data), (gst_pngdec_chain):
- * ext/libpng/gstpngdec.h:
- Added png decoder.
+2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_base_init):
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
+ (gst_tag_register_musicbrainz_tags):
+ Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
+ depend on libgsttag. This is required so we can extract/read tags like
+ DISCID without depending on libgstcddabasesrc (which used to register
+ them).
+ * gst-libs/gst/tag/gstvorbistag.c:
+ Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
+ tags (also see #347848).
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
+ Log vorbis comments we are actually writing. Const-ify array.
-2004-07-16 Julien MOUTTE <julien@moutte.net>
+2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gen_preroll_element):
+ Improve buffering a bit by avoiding a deadlock because we cannot assume
+ the underrun is always called.
-2004-07-16 Wim Taymans <wim@fluendo.com>
+2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
- * gst-libs/gst/riff/riff-read.c:
- (gst_riff_read_strf_vids_with_data),
- (gst_riff_read_strf_auds_with_data):
- * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- Make sure we don't create 0 sized subbuffers in riff-read.
- Signal the no more pads signal after reading the avi header.
-
-2004-07-16 Wim Taymans <wim@fluendo.com>
-
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_is_dynamic),
- (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
- (gst_decode_bin_init), (gst_decode_bin_dispose),
- (find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
- (no_more_pads), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (gst_decode_bin_change_state), (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (queue_overrun),
- (gen_preroll_element), (remove_prerolls), (unknown_type),
- (no_more_pads), (new_stream), (setup_source),
- (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
- (play_base_eos), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- Better error recovery. Added configurable preroll queue size. Faster
- detection of no-more-pads.
+ gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
+ Original commit message from CVS:
+ Patch by: Young-Ho Cha <ganadist at chollian dot net>
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Added MPEG-4 AAC and id and caps. Fixes #357289
+ Added WMA9 Lossless id.
-2004-07-16 Wim Taymans <wim@fluendo.com>
+2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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.
-2004-07-16 Wim Taymans <wim@fluendo.com>
+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.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- FPS seems to be 0.0 to MAX everywhere else.
+2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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)
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- mp42/mp43 (no caps) exist too.
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
- Set pixel_width/height; we've got them in-caps.
- * gst/typefind/gsttypefindfunctions.c: (plugin_init):
- * gst/wavparse/gstwavparse.c: (plugin_init):
- Both are valid primary.
- * sys/oss/gstossmixer.c:
- Remove i18n hack and enable translations.
+2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-15 Benjamin Otte <otte@gnome.org>
+ 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).
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy):
- fix for non-shm xv. Original patch by Tim Ringenbach (fixes #147248)
+2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
+ (queue_out_of_data), (gen_preroll_element),
+ (preroll_remove_overrun), (probe_triggered):
+ Refactor handling of overrun detection.
+ Separate handling of group completion and deadlock detection when doing
+ network buffering. This should fix some deadlocks that were not detected
+ because the group was completed.
+ Add more comments, improve debugging.
+
+2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/: Some more compilation fixes.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/libs/audio.c:
+ Some more compilation fixes.
+
+2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ Early morning compilation fix.
+
+2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ * configure.ac:
+ bump nano
+ Original commit message from CVS:
+ bump nano
+
+2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/check/: Fix some warnings.
+ Original commit message from CVS:
+ * tests/check/elements/gdpdepay.c: (GST_START_TEST):
+ * tests/check/elements/multifdsink.c: (GST_START_TEST):
+ * tests/check/elements/videorate.c: (GST_START_TEST):
+ * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
+ * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
+ Fix some warnings.
+
+2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_get_times):
+ change colorkey behaviour back according to #354773 comment 6/7
-2004-07-15 Benjamin Otte <otte@gnome.org>
+2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ * ChangeLog:
+ ChangeLog surgery: remove junk
+ Original commit message from CVS:
+ ChangeLog surgery: remove junk
+
+2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
+ (gst_multi_fd_sink_recover_client),
+ (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
+ (gst_multi_fd_sink_get_property):
+ * gst/tcp/gstmultifdsink.h:
+ Implement stubbed out properties unit-type, units-soft-max,
+ units-max, to allow specifying maximum sizes in units other than
+ buffers.
+ Fixes #355935
-2004-07-12 Benjamin Otte <otte@gnome.org>
+2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
+ (gst_riff_create_audio_template_caps):
+ Reorder the audio formats a bit for clarity.
+ Detect and create caps for MSGSM and MSN (WAV49).
+ Fixes #356596.
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
+ Small cleanups, move error handling out of normal flow for clarity.
+
+2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add new interface to control video orientation (fixes #354908)
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs.types:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/videoorientation.c:
+ (gst_video_orientation_get_type),
+ (gst_video_orientation_iface_init),
+ (gst_video_orientation_get_hflip),
+ (gst_video_orientation_get_vflip),
+ (gst_video_orientation_get_hcenter),
+ (gst_video_orientation_get_vcenter),
+ (gst_video_orientation_set_hflip),
+ (gst_video_orientation_set_vflip),
+ (gst_video_orientation_set_hcenter),
+ (gst_video_orientation_set_vcenter):
+ * gst-libs/gst/interfaces/videoorientation.h:
+ Add new interface to control video orientation (fixes #354908)
+
+2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ Use G_UNLIKELY in _create and log one more detail.
+ (gst_video_test_src_get_times), (gst_video_test_src_create):
+ * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
+ Use gst_util_uint64_scale_int in _get_times().
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_alloc_buffer):
- use bufferpools
+2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
+ Give better warning message (add object and detail).
- * 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.
+2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-13 Johan Dahlin <johan@gnome.org>
+ sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
+ (gst_xvimagesink_get_times):
+ xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
+ #354773), use gst_util_uint64_scale_int in _get_times()
- * gst/playback/gstplaybin.c (gst_play_bin_set_property)
- (gst_play_bin_get_property): Impl.
+2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
-2004-07-13 Wim Taymans <wim@fluendo.com>
+ ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
+ Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
+ always true, leading to dropping all timestamps.
- * ext/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.
+2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-07-13 Wim Taymans <wim@fluendo.com>
+ ext/libvisual/visual.c: update to work also with libvisual 0.4 API
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_vis_src_negotiate),
+ (gst_visual_chain), (gst_visual_change_state):
+ update to work also with libvisual 0.4 API
+ * tools/gst-launch-ext.1.in:
+ * tools/gst-visualise.1.in:
+ remove references to old man-pages
+ * tests/examples/seek/seek.c: (main):
+ add real meadi-buttons, add tool-tips for the seek-options, arrange
+ seek options in a table
- * gst-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.
+2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
+ (gst_ogg_mux_push_buffer):
+ Don't generate out-of-order timestamps from oggmux, instead clamp
+ output timestamps to be >= the previously output ts.
+ Fixes #355595
- * configure.ac:
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_dispose), (gst_play_set_location),
- (gst_play_set_data_src), (gst_play_set_video_sink),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_sink_element),
- (gst_play_get_all_by_interface):
- * gst-libs/gst/play/play.h:
- add new method to get elements implementing an interface.
- add various error logging
-
-2004-07-08 Wim Taymans <wim@fluendo.com>
-
- * examples/seeking/seek.c: (make_dv_pipeline), (make_avi_pipeline),
- (make_mpeg_pipeline), (make_mpegnt_pipeline),
- (make_playerbin_pipeline), (query_durations_elems),
- (query_durations_pads), (query_positions_elems),
- (query_positions_pads), (update_scale), (iterate), (stop_seek),
- (main):
- Added playbin seeking example.
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * gst-libs/gst/play/play.c: (gst_play_set_location),
- (gst_play_set_data_src), (gst_play_set_video_sink),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_framerate):
- use a macro to look up elements from hash table
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_get_length_callback), (gst_play_set_location),
- (gst_play_seek_to_time), (gst_play_set_data_src),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization), (gst_play_connect_visualization),
- (gst_play_get_sink_element):
- - add debugging info
- - fix looking up sink elements by iterating over complete caps
- - put everything except for source and autoplugger in a complete bin
-
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_drain_audio):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_flush_one_pad),
- (gst_alsa_sink_check_event), (gst_alsa_sink_mmap),
- (gst_alsa_sink_write), (gst_alsa_sink_loop):
- * ext/alsa/gstalsasink.h:
- - add debugging info
- - clean up schizophrenia of data/buffer/event
- - fix double event unref error
+2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
-2004-07-08 Wim Taymans <wim@fluendo.com>
+ gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_multi_fd_sink_class_init):
+ Updates, fixes, and typo corrections for multifdsink. No functional
+ changes.
- * gst/playback/Makefile.am:
- Add headers to noinst
+2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
-2004-07-08 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
+ Don't crash on truncated files - check that we got an 8 byte buffer
+ before trying to memcmp it.
- * tools/gst-launch-ext-m.m:
- * tools/gst-launch-ext.1.in:
- convert to the third millenium
+2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 David Schleef <ds@schleef.org>
+ gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (get_active_source):
+ Make stream-switching appear instant to the application
+ (ie. make sure that a g_object_get on 'current-foo' returns
+ the stream previously set with g_object_set(). Totem needs
+ this to update stream-related meta-info (like audio-codec)
+ correctly when switching streams.
- * sys/dxr3/Makefile.am: noinst_SOURCES should be nodist_SOURCES
+2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/playback/Makefile.am:
- * gst/playback/README:
- * gst/playback/gstdecodebin.c: (gst_decode_bin_get_type),
- (gst_decode_bin_class_init), (gst_decode_bin_factory_filter),
- (compare_ranks), (print_feature), (gst_decode_bin_init),
- (gst_decode_bin_dispose), (find_compatibles), (close_pad_link),
- (try_to_link_1), (new_pad), (close_link), (type_found),
- (gst_decode_bin_set_property), (gst_decode_bin_get_property),
- (plugin_init):
- * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_type),
- (gst_play_base_bin_class_init), (gst_play_base_bin_init),
- (gst_play_base_bin_dispose), (rebuild_pipeline), (queue_overrun),
- (gen_preroll_element), (no_more_pads), (new_stream),
- (setup_source), (gst_play_base_bin_set_property),
- (gst_play_base_bin_get_property), (gst_play_base_bin_change_state),
- (gst_play_base_bin_add_element),
- (gst_play_base_bin_remove_element),
- (gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream),
- (gst_play_base_bin_unlink_stream),
- (gst_play_base_bin_get_streaminfo):
- * gst/playback/gstplaybasebin.h:
- * gst/playback/gstplaybin.c: (gst_play_bin_get_type),
- (gst_play_bin_class_init), (gst_play_bin_init),
- (gst_play_bin_dispose), (gst_play_bin_set_property),
- (gst_play_bin_get_property), (gen_video_element),
- (gen_audio_element), (setup_sinks), (gst_play_bin_change_state),
- (gst_play_bin_get_event_masks), (gst_play_bin_send_event),
- (gst_play_bin_get_formats), (gst_play_bin_convert),
- (gst_play_bin_get_query_types), (gst_play_bin_query),
- (plugin_init):
- * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
- (gst_stream_info_get_type), (gst_stream_info_class_init),
- (gst_stream_info_init), (gst_stream_info_new),
- (gst_stream_info_dispose), (gst_stream_info_set_property),
- (gst_stream_info_get_property):
- * gst/playback/gststreaminfo.h:
- * gst/playback/test.c: (gen_video_element), (gen_audio_element),
- (main):
- * gst/playback/test2.c: (main):
- * gst/playback/test3.c: (update_scale), (main):
- More playbin fixes. Added README. Do better element filtering.
- Added base class to preroll media. Added test apps.
+2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
+ Original commit message from CVS:
+ * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
+ (gst_audio_convert_transform_caps):
+ Get structure-name just once.
+
+2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ tests/check/: Fix big batch of compiler warnings.
+ Original commit message from CVS:
+ * tests/check/elements/audioresample.c: (GST_START_TEST):
+ * tests/check/elements/videotestsrc.c: (check_rgb_buf):
+ * tests/check/elements/volume.c: (GST_START_TEST):
+ * tests/check/elements/vorbisdec.c: (GST_START_TEST):
+ * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
+ (test_pipeline), (GST_START_TEST):
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
+ Fix big batch of compiler warnings.
+
+2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Add docs about icydemux usage in connection with gnomevfssrc
+ * ext/libvisual/visual.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c:
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst/audiorate/gstaudiorate.c:
+ More G_OBJECT macro fixing.
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Fix wrong info in header due to copy & paste
+
+2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
+ (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
+ (gst_base_audio_src_create), (gst_base_audio_src_change_state):
+ Do the delay calculation in the source/sink base classes as this is
+ specific for the capture/playback mode.
+ Try to fixate a bit better, like round depth up to a multiple of 8
+ bigger than width.
+ Handle underruns correctly by marking DISCONT on buffers and adjusting
+ timestamps to handle the gap.
+ Set offset/offset_end correctly on buffers.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
+ (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
+ (gst_ring_buffer_read):
+ Remove resync and underrun recovery from the ringbuffer.
+ Fix ringbuffer read code on under/overrun.
+
+2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
+ (gst_play_base_bin_init), (fill_buffer), (check_queue),
+ (queue_threshold_reached), (gst_play_base_bin_set_property),
+ (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybasebin.h:
+ Don't use a 0 low watermark when buffering, it is catching starvation
+ way too late. Instead, use a 3 second queue with 30 and 95
+ percent low/high watermarks.
+ Added queue-min-threshold property to configure low watermark.
+ Use new _buffering message API.
+ Make queue_threshold variable big enough to store a uint64 time value.
+ API: playbin::queue-min-threshold property.
+
+2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ configure.ac: We require 0.10.10.1 now because of _wait_preroll().
+ Original commit message from CVS:
+ * configure.ac:
+ We require 0.10.10.1 now because of _wait_preroll().
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Use gst_base_sink_wait_preroll().
+
+2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/: Use DEBUG_OBJECT more.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
+ * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
+ Use DEBUG_OBJECT more.
-2004-07-07 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+=== release 0.10.10 ===
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_flush_decoder):
- * ext/mpeg2dec/gstmpeg2dec.h:
- various debugging improvements. Reset stream to next picture
- instead of sequence header, otherwise seeks cannot work.
+2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gdp.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * ext/theora/theoraparse.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst/playback/gstplaybin.c:
+ * tests/check/Makefile.am:
+ * win32/common/config.h:
+ releasing 0.10.10
+ Original commit message from CVS:
+ releasing 0.10.10
- * gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type),
- (gst_video_box_class_init), (gst_video_box_set_property),
- (gst_video_box_i420), (gst_video_box_ayuv), (gst_video_box_chain):
- Use pad_alloc where possible.
+2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ * configure.ac:
+ * win32/common/config.h:
+ second prerelease
+ Original commit message from CVS:
+ second prerelease
- * sys/oss/gstosselement.c: (gst_osselement_reset),
- (gst_osselement_parse_caps):
- * sys/oss/gstosselement.h:
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- Fix offset on osssrc.
+2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-07-07 Wim Taymans <wim@fluendo.com>
+ * ChangeLog:
+ update bug in changelog
+ Original commit message from CVS:
+ update bug in changelog
- * 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.
+2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
-2004-07-06 David Schleef <ds@schleef.org>
+ Fix implementation of sync-method 'next-keyframe'
+ Original commit message from CVS:
+ patch by: Michael Smith <msmith at fluendo dot com>
+ * gst/tcp/gstmultifdsink.c: (is_sync_frame),
+ (gst_multi_fd_sink_client_queue_buffer),
+ (gst_multi_fd_sink_new_client):
+ * tests/check/elements/multifdsink.c: (GST_START_TEST),
+ (multifdsink_suite):
+ Fix implementation of sync-method 'next-keyframe'
- * gst/playback/Makefile.am: 'test' in bin_PROGRAMS? Are you
- serious? (Fixed, obviously.)
+2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
-2004-07-06 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * 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>
+2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
* 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.
+ * win32/common/config.h:
+ first prerelease
+ Original commit message from CVS:
+ first prerelease
-2004-07-02 Benjamin Otte <otte@gnome.org>
+2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.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.
+ * 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
-2004-06-29 Wim Taymans <wim@fluendo.com>
+2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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.
-2004-06-26 Wim Taymans <wim@fluendo.com>
+2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/Makefile.am:
- * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
- (gst_multifdsink_get_type), (gst_multifdsink_base_init),
- (gst_multifdsink_class_init), (gst_multifdsink_init),
- (gst_multifdsink_debug_fdset), (gst_multifdsink_client_remove),
- (gst_multifdsink_handle_client_read),
- (gst_multifdsink_client_queue_data),
- (gst_multifdsink_client_queue_caps),
- (gst_multifdsink_client_queue_buffer),
- (gst_multifdsink_handle_client_write),
- (gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
- (gst_multifdsink_handle_clients), (gst_multifdsink_thread),
- (gst_multifdsink_chain), (gst_multifdsink_set_property),
- (gst_multifdsink_get_property), (gst_multifdsink_init_send),
- (gst_multifdsink_close), (gst_multifdsink_change_state):
- * gst/tcp/gstmultifdsink.h:
- * gst/tcp/gsttcpplugin.c: (plugin_init):
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_get_type),
- (gst_tcpserversink_class_init), (gst_tcpserversink_init),
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_handle_select),
- (gst_tcpserversink_set_property), (gst_tcpserversink_get_property),
- (gst_tcpserversink_init_send), (gst_tcpserversink_close):
- * gst/tcp/gsttcpserversink.h:
- Added multifdsink, made tcpserversink a subclass of fdsink, removed
- one of the locks, added recovery policy to multifdsink.
-
-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
+ 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).
-2004-06-26 Thomas Vander Stichele <thomas at apestaart dot org>
+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/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.
+2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
-2004-06-24 Wim Taymans <wim@fluendo.com>
+ 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
- * gst/audiorate/gstaudiorate.c: (gst_audiorate_link),
- (gst_audiorate_init), (gst_audiorate_chain),
- (gst_audiorate_set_property), (gst_audiorate_get_property):
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_chain):
- Added some logging, fixed an overflow bug in videorate.
+2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-24 Benjamin Otte <otte@gnome.org>
+ 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.
- * ext/kio/Makefile.am:
- fix for builddir != srcdir and distcheck
+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.
-2004-06-24 Benjamin Otte <otte@gnome.org>
+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").
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- * gst/tcp/Makefile.am:
- * sys/dxr3/Makefile.am:
- don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
- tarball.
- Also add all *.list files that were missing.
- * Makefile.am:
- add a distcheck hook to ensure the above doesn't happen again.
+2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-23 David I. Lehn <dlehn@users.sourceforge.net>
+ 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).
- * ext/Makefile.am: s/DTS_DIR=dvdread/DTS_DIR=dts/
+2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-23 Colin Walters <walters@redhat.com>
+ gst/adder/gstadder.c: Fix build.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_src_event):
+ Fix build.
- * m4/Makefile.am: Distribute gst-fionread.m4.
+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).
-2004-06-23 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * configure.ac: back to dev
+2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-23 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
-=== release 0.8.2 ===
+ ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
+ (gst_vorbis_enc_buffer_check_discontinuous),
+ (gst_vorbis_enc_chain):
+ Ignore explicit DISCONT marked on buffers (which is often spurious,
+ particularly when using multiple segments), in favour of solely
+ using the timestamps/durations.
-2004-06-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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
+ gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
+ Don't rely on incoming buffers offset anymore, since it is completely
+ broken when using multiple segments.
+ Instead convert the incoming buffers timestamp to running time, and
+ then convert that value to the offsets.
+ Also inform GstSegment of the last outputted stop position, which is
+ needed if we received several segments with an unknown stop value.
-2004-06-23 Wim Taymans <wim@fluendo.com>
+2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
+ fix buffer unreffing on a header push failure
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.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.
+ gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
+ (gst_audio_rate_chain):
+ Make the metadata of the buffer writable before changing its
+ flags.
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.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.
+ * ChangeLog:
+ Fix changelog with bugzilla bug it fixed.
+ Original commit message from CVS:
+ Fix changelog with bugzilla bug it fixed.
+
+2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/audiorate/gstaudiorate.c: Fix audiorate some more.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
+ (gst_audio_rate_setcaps), (gst_audio_rate_init),
+ (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
+ (gst_audio_rate_chain), (gst_audio_rate_change_state):
+ Fix audiorate some more.
+ Reset and resync counters on flush and READY.
+ Handle the DISCONT flag correctly.
+ Use GstSegment to track position.
+ Fail when not negotiated.
+
+2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
+
+ gst/tcp/gstmultifdsink.c: Fix spelling.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
+ Fix spelling.
+ Remove accidently included debug line.
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_init):
- Initialize the property as well.
+ 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.
-2004-06-22 Wim Taymans <wim@fluendo.com>
+2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
- * gst/videorate/gstvideorate.c: (gst_videorate_class_init),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_set_property), (gst_videorate_get_property):
- Add property to make videorate silent.
- Add property to prefer new frames over old ones.
+ gst/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.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
- * sys/osxvideo/Makefile.am:
- Workaround so that the osxvideo .so file gets linked with the
- Cocoa, OpenGL and QuickTime frameworks
+ ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c:
+ (gst_vorbis_enc_buffer_check_discontinuous):
+ Allow very small discontinuities in the timestamps. These we can't
+ do anything useful with anyway (because vorbis's timestamps have
+ only sample granularity), and are commonly produced by elements with
+ minor bugs. Allow up to 1/2 a sample out.
+ Fixes #351742.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/osxaudio/Makefile.am:
- Workaround so that the osxaudio .so file gets linked with the
- CoreAudio framework
+ tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
+ (play_scrub_toggle_cb), (main):
+ Add a checkbox to enable play scrubbing. Makes it possible to disable
+ normal scrubbing.
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac:
- Whoops, my fault...fixed build issues
+ tests/check/elements/.cvsignore: make buildbot happy
+ Original commit message from CVS:
+ * tests/check/elements/.cvsignore:
+ make buildbot happy
-2004-06-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
+ (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
+ (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
+ (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
+ (gst_ogm_text_parse_strip_trailing_zeroes),
+ (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
+ (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
+ Refactor ogm parse, do better input checking, misc. clean-ups.
+ Cache incoming events and push them once the source pad has
+ been created. Don't pass unterminated strings to sscanf().
+ Strip trailing zeroes from subtitle text output, since they
+ are not valid UTF-8. Don't push vorbiscomment packets on
+ the subtitle text pad. Output perfect streams if possible.
- * 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.
+2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-06-18 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.
- * ext/sidplay/gstsiddec.cc:
- Fix negotiation and set correct end offset.
+2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
+ Original commit message from CVS:
+ * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
+ Unit test fixes: \377 is more likely to fit into 8 bits than \777;
+ actually return return value in taglists_are_equal.
- * configure.ac: branch and prerelease
+2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/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
+2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
+ (gst_riff_parse_chunk), (gst_riff_parse_file_header),
+ (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
+ (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
+ (gst_riff_parse_info):
+ Protect public functions against bad input.
+ Do some cleanups.
+ Fix documentation.
-2004-06-17 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Add voxware audio IDs (even if we can't play it) (#351795).
-2004-06-17 Wim Taymans <wim@fluendo.com>
+2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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-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.
- * gst/videorate/Makefile.am:
- * gst/videorate/gstvideorate.c: (gst_videorate_get_type),
- (gst_videorate_base_init), (gst_videorate_class_init),
- (gst_videorate_getcaps), (gst_videorate_link),
- (gst_videorate_init), (gst_videorate_chain),
- (gst_videorate_set_property), (gst_videorate_get_property),
- (gst_videorate_change_state), (plugin_init):
- Added a video timestamp corrector.
+2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ (gst_tag_list_to_vorbiscomment_buffer):
+ * tests/check/libs/tag.c: (GST_START_TEST):
+ And the same for _to_vorbiscomment_buffer(): allow
+ id_data_len == 0 for speex.
- fixed a potential leak with previous commit
+2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
+ * gst/gdp/README:
+ adding a README
+ Original commit message from CVS:
+ adding a README
-2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
- Added missing refcount, fixes bug #144425
- Cheers Tim for finding the bug
+ 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.
-2004-06-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/gstv4lcolorbalance.c:
- * sys/v4l/gstv4lcolorbalance.h:
- * sys/v4l/gstv4lelement.c:
- * sys/v4l/gstv4lelement.h:
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
- * sys/v4l/gstv4lmjpegsink.h:
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
- * sys/v4l/gstv4lmjpegsrc.h:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_fps_list),
- (gst_v4lsrc_get_fps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_get):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/gstv4ltuner.c:
- * sys/v4l/gstv4ltuner.h:
- * sys/v4l/gstv4lxoverlay.c:
- * sys/v4l/gstv4lxoverlay.h:
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay),
- (gst_v4l_set_window), (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_set_window_properties),
- (gst_v4l_open), (gst_v4l_get_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_sync_frame),
- (gst_v4lmjpegsink_set_buffer), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start), (gst_v4lmjpegsink_get_buffer),
- (gst_v4lmjpegsink_play_frame), (gst_v4lmjpegsink_wait_frame),
- (gst_v4lmjpegsink_playback_stop),
- (gst_v4lmjpegsink_playback_deinit):
- * sys/v4l/v4lmjpegsink_calls.h:
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame),
- (gst_v4lmjpegsrc_sync_next_frame), (gst_v4lmjpegsrc_set_buffer),
- (gst_v4lmjpegsrc_set_capture), (gst_v4lmjpegsrc_set_capture_m),
- (gst_v4lmjpegsrc_capture_init), (gst_v4lmjpegsrc_capture_start),
- (gst_v4lmjpegsrc_grab_frame), (gst_v4lmjpegsrc_requeue_frame),
- (gst_v4lmjpegsrc_capture_stop), (gst_v4lmjpegsrc_capture_deinit):
- * sys/v4l/v4lmjpegsrc_calls.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_set_capture),
- (gst_v4lsrc_capture_init), (gst_v4lsrc_capture_start),
- (gst_v4lsrc_grab_frame), (gst_v4lsrc_requeue_frame),
- (gst_v4lsrc_capture_stop), (gst_v4lsrc_capture_deinit),
- (gst_v4lsrc_try_palette):
- * sys/v4l/v4lsrc_calls.h:
- bunch of paranoia cleanups
+ 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().
-2004-06-14 David Schleef <ds@schleef.org>
+2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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:
+ 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).
-2004-06-14 Benjamin Otte <otte@gnome.org>
+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
- * ext/alsa/gstalsa.c: Use snd_pcm_hw_params_set_rate _near instead of
- snd_pcm_hw_params_set_rate since the latter fails for no good
- reason on some setups.
+2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-06-14 David Schleef <ds@schleef.org>
+ 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
- * gst/volume/demo.c: (value_changed_callback): exp10() is not
- standard. Thank you for playing.
+2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-14 Wim Taymans <wim@fluendo.com>
+ * 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.
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
- Patch 1.3 broke the ordering of the colorspace info and
- made the plugin basically work by coincidence, reordered
- the info.
+2006-08-17 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.
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ 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.
-2004-06-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/volume/.cvsignore:
- * gst/volume/Makefile.am:
- * gst/volume/demo.c: (value_changed_callback), (idler),
- (setup_gui), (main):
- added small demo app
+ 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().
-2004-06-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/esd/esdsink.c: (gst_esdsink_change_state):
- * ext/esd/esdsink.h:
- Close the esd connection on pause, because esd will just wait -
- blocking all other esd clients indefinitely.
+2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Christophe Fergeau <teuf@gnome.org>
+ 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).
- * gst/tags/gstvorbistag.c: replaced a g_warning which I added in my
- previous commit with GST_DEBUG
+2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac:
- add a header check for a dvdread header in dvdnav. Fixes #133002
+2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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).
- * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init_send):
- * gst/tcp/gsttcpclientsink.h:
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init_receive):
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init),
- (gst_tcpserversink_handle_server_read),
- (gst_tcpserversink_init_send):
- * gst/tcp/gsttcpserversink.h:
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
- * gst/tcp/gsttcpserversrc.h:
- Modified the tcp plugins so they are portable (IPv4,IPv6, any future
- version of IP)
+2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-06-12 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+ 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.
- * configure.ac:
- Added ogg library so that OSX detects libtheora properly
-
-2004-06-11 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoradec.c: (theora_dec_chain),
- (theora_dec_change_state):
- Don't try to decode frames before we received a keyframe.
-
-2004-06-11 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_init), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property):
- Added property to set the maximum delay of a page.
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
- (gst_ogg_mux_init), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property):
- Added max-delay property to control the maximum amount
- of data to put in one page.
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (gst_theora_enc_init), (theora_enc_sink_link),
- (theora_buffer_from_packet), (theora_enc_set_property),
- (theora_enc_get_property):
- Set duration on encoded buffer, added some more properties
-
-2004-06-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_get_headers), (gst_ogg_mux_set_header_on_caps),
- (gst_ogg_mux_send_headers), (gst_ogg_mux_loop):
- * ext/theora/theoraenc.c: (theora_enc_chain):
- Fix refcounting bugs
-
-2004-06-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init),
- (gst_asf_demux_loop), (gst_asf_demux_process_file),
- (gst_asf_demux_process_data), (gst_asf_demux_handle_data),
- (gst_asf_demux_process_object), (gst_asf_demux_get_stream),
- (gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event),
- (gst_asf_demux_handle_src_event), (gst_asf_demux_handle_src_query),
- (gst_asf_demux_change_state):
- * gst/asfdemux/gstasfdemux.h:
- You know Chimaira? "I - HATE - EVERYTHING". Yeah, that's what this
- feels like. I think we should set a new requirement for demuxers
- from now on to implement sane loop functions, data loops, query
- and seek functions before first commit into CVS. And this commit
- fixes all of the above.
-
-2004-06-10 Christophe Fergeau <teuf@gnome.org>
-
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add): make sure parsed
- vorbis comments are properly encoded in UTF-8 before adding them
- to a GstTagList
-
-2004-06-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (add_channels):
- handle min <= max correctly
- * ext/alsa/gstalsa.c: (gst_alsa_fixate_to_mimetype),
- (gst_alsa_fixate_field_nearest_int), (gst_alsa_fixate):
- add fixation functions so we fixate correctly. No preferring of alaw
- anymore because it's the first structure.
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsa.c: (gst_alsa_sw_params_dump),
- (gst_alsa_hw_params_dump):
- add functions to ease debugging in alsalib
- * ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
- (gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
- (gst_alsa_start_audio):
- only specify hw params if we really setup a format (fixes #134007 -
- or at least works around it)
-
-2004-06-09 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_init),
- (gst_ogg_mux_next_buffer), (gst_ogg_mux_buffer_from_page),
- (gst_ogg_mux_push_page), (gst_ogg_mux_get_headers),
- (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_send_headers),
- (gst_ogg_mux_loop):
- Use stream caps to setup the initial pages in the ogg stream.
- Correctly set the streamheader caps on the srcpad.
-
-2004-06-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_fps_list),
- (gst_v4lsrc_get_fps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps):
- * sys/v4l/v4l_calls.c: (gst_v4l_set_window_properties),
- (gst_v4l_get_picture), (gst_v4l_get_audio), (gst_v4l_set_audio):
- add querying of fps lists for webcams. Negotiating to a framerate
- now works.
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/theora/theoraenc.c: (theora_buffer_from_packet),
- (theora_push_buffer), (theora_push_packet),
- (theora_set_header_on_caps), (theora_enc_chain):
- mark buffers and put on streamheader, raw theora streaming
- now works too, whee
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_header),
- (gst_tcp_gdp_read_caps):
- do a looping read for caps and GDP headers too
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get):
- return EOS instead of NULL in _get
-
-2004-06-08 Wim Taymans <wim@fluendo.com>
-
- * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_header),
- (gst_tcp_gdp_read_caps), (gst_tcp_gdp_write_header),
- (gst_tcp_gdp_write_caps):
- * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get):
- * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_gdp_read_caps),
- (gst_tcpserversrc_gdp_read_header), (gst_tcpserversrc_get):
- Memory leak fixes
-
-2004-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-14 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.
- * 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>
+2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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>
+ 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).
- * AUTHORS:
- Add me to the authors file
- * configure.ac:
- Increase the libdv requirement to >= version 0.100
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_src_convert),
- (gst_dvdec_src_query), (gst_dvdec_handle_sink_event),
- (gst_dvdec_push), (gst_dvdec_loop), (gst_dvdec_change_state):
- * ext/dv/gstdvdec.h:
- Add support for the new_media flag when sending DISCONT events
- Make the querying work when video pad is not linked
-
-2004-06-07 Tim-Philipp Müller <t.i.m@zen.co.uk>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init):
- create a NULL-initialized array of pads, so we don't think they
- exist already. (fixes #143130)
-
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_init),
- (mixmatrix_resize), (gst_mixmatrix_set_all_caps),
- (gst_mixmatrix_request_new_pad), (gst_mixmatrix_loop):
- don't use // coments
-
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_samples_to_timestamp):
- cast to GstClockTime to get higher granularity
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- use gst_element_set_time_delay to get the exact time
- * ext/mad/gstmad.c: (gst_mad_chain):
- use the negotiated rate instead of the current frame's rate which
- might be wrong because of bit errors. This avoids emitting totally
- bogus timestamps and screwing sync.
- (fixes #143454)
-
-2004-06-07 Tim-Philipp Müller <t.i.m@zen.co.uk>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * gst/adder/gstadder.c: (gst_adder_loop):
- properly error out when no negotiation has happened yet. (fixes
- #143032)
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- forward correctly transformed offset in discont events. Based on
- patch by Arwed v. Merkatz. (fixes #142851)
-
-2004-06-06 David Schleef <ds@schleef.org>
-
- * gst/ffmpegcolorspace/gstffmpegcodecmap.c: that's
- G_HAVE_GNUC_VARARGS, not G_HAVE_GNU_VARARGS. Should fix compile
- problems on several systems.
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init):
- use explicit caps on the srcpad
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- properly error out if caps couldn't be set (fixes #142764)
-
-2004-06-06 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsa.c: (gst_alsa_probe_hw_params),
- (gst_alsa_set_hw_params), (gst_alsa_set_sw_params),
- (gst_alsa_start_audio):
- - don't call set_periods_integer anymore, it breaks the
- configuration randomly
- - call snd_pcm_hw_params_set_access directly instead of using masks
- - don't fail if the sw_params can't be set, just use the default
- params and hope it works. Alsalib has weird issues when you touch
- sw_params and does no proper error reporting about what failed.
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio),
- (gst_alsa_close_audio):
- make our alsa debugging go via gst debugging and not conditionally
- defined
- * ext/alsa/gstalsa.h:
- add ALSA_DEBUG_FLUSH macro
- * ext/alsa/gstalsaplugin.c: (gst_alsa_error_wrapper),
- (plugin_init):
- wrap alsa errors to be printed via the gst debugging system and not
- spammed to stderr
-
-2004-06-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_init),
- (gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event),
- (gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header), (qtdemux_dump_mvhd),
- (qtdemux_parse_trak):
- * gst/qtdemux/qtdemux.h:
- Bitch. Also known as seeking, querying & co.
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- * sys/oss/gstosssink.h:
- Resyncing is for weenies, this hack is no longer needed and was
- broken anyway (since it - unintendedly - always leaves resync to
- TRUE).
-
-2004-06-05 Andrew Turner <zxombie@hotpop.com>
-
- * gst/tcp/gsttcp.c: portability (Solaris 10/FreeBSD)
- * gst/tcp/gsttcpclientsrc.h: idem
- - define MSG_NOSIGNAL if not done
- - include unistd.h for off_t
- (fixes #143749)
-
-2004-06-05 Benjamin Otte <otte@gnome.org>
+2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/kio/Makefile.am:
- check for qt's moc preprocessor explicitly and use it
+ gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
+ The checks here are not even close to anything that would
+ justify MAXIMUM probability, lowering to POSSIBLE until someone
+ fixes the checks (case at hand: quicktime redirection files
+ might start with 00 00 01 XX and pass the checks here just
+ fine, see #350399).
-2004-06-03 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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)
+ 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"
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_get_type):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- check error condition on available samples correctly
+ 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.
-2004-06-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
- * 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
+ gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
+ Better detection for multipart/x-mixed-replace: accept leading
+ whitespaces before the boundary marker as well (as our very own
+ multipartmux used to produce) (#349068).
-2004-06-02 Wim Taymans <wim@fluendo.com>
+2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
- reviewed by: Johan
+ 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).
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_loop):
- Oops
+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.
-2004-06-02 Wim Taymans <wim@fluendo.com>
+2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
- (gst_multipart_mux_init), (gst_multipart_mux_loop),
- (gst_multipart_mux_get_property), (gst_multipart_mux_set_property),
- (gst_multipart_mux_change_state):
- Added configurable boundary specifier, added the value as a
- caps field as well.
+ 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).
-2004-06-02 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/tcp/gsttcp.c:
- * gst/tcp/gsttcpclientsrc.c:
- * gst/tcp/gsttcpclientsrc.h:
- * gst/tcp/gsttcpserversrc.c:
- - portability fix, to compile on OSX
- (fixes #143146)
+ gst/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).
- * sys/osxaudio/gstosxaudioelement.c:
- * sys/osxaudio/gstosxaudiosink.c:
- * sys/osxaudio/gstosxaudiosrc.c:
- - compilation warnings on OSX
- (fixes #143153)
+2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
-2004-06-02 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
+ Original commit message from CVS:
+ Patch by: Jens Granseuer <jensgr at gmx net>
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ Add typefinder for Interplay's MVE format (#348973).
- * ext/vorbis/vorbisdec.c : sign warning fixes
+2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
+ Original commit message from CVS:
+ Patch by: Marcel Moreaux <marcelm at luon dot net>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_add_to_queue):
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ Handle RTP sequence number rollover.
+ Disable jitterbuffer by default.
+
+2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/gdp/gstgdpdepay.c: Disable seeking.
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
+ (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
+ (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
+ (gst_gdp_depay_change_state):
+ Disable seeking.
+ Small cleanups.
+ Clear adapter on disconts.
+ Clear caps when going to READY instead of NULL
+ * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
+ (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
+ (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
+ (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
+ (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
+ (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
+ (gst_gdp_pay_change_state):
+ * gst/gdp/gstgdppay.h:
+ Reset payloader when going to READY.
+ Fix leaked buffers in ->queue on push errors.
+ Disable seeking.
+ Code cleanups.
+ Create packetizer in _init, free in _finalize.
+
+2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
+ Original commit message from CVS:
+ * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
+ (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
+ Consume all events except EOS because we generate events from
+ the gdp payload instead. Fixes #349204
+
+2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c: (audioresample_stop),
+ (audioresample_set_caps):
+ Don't leak references to the incoming caps. Clean them up when
+ stopping.
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
+ (gst_video_scale_finalize):
+ Don't leak our temporary pixel buffer.
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
+ (GST_START_TEST), (simple_launch_lines_suite):
+ Fix leaks and re-enable the test for valgrind checking.
+
+2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
+
+ gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
+ Original commit message from CVS:
+ Patch by: Sjoerd Simons <sjoerd at luon net>
+ * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
+ (plugin_init):
+ Add typefind function for multipart/x-mixed-replace (#348916).
- * gst-libs/gst/mixer/mixertrack.c :
- do no use defines which are glib 2.4 specific
+2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-06-01 Christophe Fergeau <teuf@gnome.org>
+ gst/adder/gstadder.c: Fix leak in duration query.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_setcaps),
+ (gst_adder_query_duration):
+ Fix leak in duration query.
+ Reflow some docs and notes.
- * ext/flac/gstflactag.c: strip ending framing bit from vorbiscomment
- buffer since libflac doesn't expect it (reports a sync error when
- it encounters that)
+2006-07-28 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.
-2004-06-01 Owen Fraser-Green <owen@discobabe.net>
+2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
- * 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
+ 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.
+2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
+
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
+ Original commit message from CVS:
+ patch by: Kai Vehmanen <kv2004 eca cx>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_chain),
+ (gst_base_rtp_depayload_handle_sink_event),
+ (gst_base_rtp_depayload_change_state):
+ Don't send multiple newsegments with different formats.
+ Fixes #348677.
+
+2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
+ (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
+ Make seeking in ogg more accurate again by doing the more correct
+ granuletime to stream time conversion.
+
+2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_new_client):
+ debug a little more understandably
+ do not use goto as a substitute for break, especially if
+ break is also being used
-2004-06-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio):
- improve error messages on open
+ * 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
+2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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.
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- check if v4l-conf is in path
+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.
-2004-06-01 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * gst-libs/gst/media-info/media-info-priv.c: (gmi_set_mime):
- change assert to a more readable error message
+2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-31 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * gst-libs/gst/tuner/tunerchannel.h:
- - add a freq_multiplicator field to make the conversion
- between internal frequency unit and Hz
- * sys/v4l/gstv4lelement.c:
- * sys/v4l2/gstv4l2element.c:
- - change default video device to /dev/video0
- * sys/v4l/v4l_calls.c:
- * sys/v4l2/v4l2_calls.c:
- - we only expose frequency to the user in Hz instead of
- bastard v4lX unit (either 62.5kHz or 62.5Hz)
+2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-31 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- Initialise b_o_s and e_o_s variables
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add some unusual fourcc's from mplayer avi's
- * gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
- Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
- autoplugging.
+ gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add 'fact' chunk id
-2004-05-28 Wim Taymans <wim@fluendo.com>
+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).
- * 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>
+2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.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-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/audioconvert/gstaudioconvert.c:
- (gst_audio_convert_parse_caps):
- make sure we don't allow depth > width
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
- fixate endianness to G_BYTE_ORDER as default
- * gst/audioscale/gstaudioscale.c:
- we don't handle another endianness as host-endianness
+2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-05-25 David Schleef <ds@schleef.org>
+ gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c:
+ (gst_riff_create_video_template_caps):
+ Add more FOURCCs (sort list to make stuff easier to find),
+ add comment what those 16 bytes in struct _gst_riff_strh according to
+ one avi-dumper are
- * gst/ffmpegcolorspace/mem.c: malloc() is in stdlib.h, not malloc.h
+2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-24 Benjamin Otte <otte@gnome.org>
+ 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
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
- (gst_oggvorbisenc_setup):
- properly fail when we can't setup the vorbis encoder due to
- unsupported settings
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
- (gst_vorbisenc_setup):
- same
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- fix case where warnings occured when one pad was unlinked while the
- other's link function was called
+2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * gst/tcp/Makefile.am:
- use GST_ENABLE_NEW
+2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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).
- * gst-libs/gst/resample/private.h:
- don't use optimizations that are #if 0'ed
+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
-2004-05-24 Wim Taymans <wim@fluendo.com>
+=== release 0.10.9 ===
- * 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.
+2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-05-24 Wim Taymans <wim@fluendo.com>
+ 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"
- * gst/videoscale/videoscale.c: (gst_videoscale_scale_nearest),
- (gst_videoscale_scale_nearest_str2),
- (gst_videoscale_scale_nearest_str4),
- (gst_videoscale_scale_nearest_32bit),
- (gst_videoscale_scale_nearest_24bit),
- (gst_videoscale_scale_nearest_16bit):
- Fix the scaling algorithm and avoid a buffer overflow.
- removed the while loop in the scaling function as it
- was used for point sampling only.
+2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
-2004-05-24 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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.
- * 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
+2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-24 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ * 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>
- * ext/Makefile.am:
- add rules to build shout2send (was removed by accident
- when this module was no more marked experimental/broken)
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-05-24 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/shout2/gstshout2.c:
- * ext/shout2/gstshout2.h:
- adding a "connection problem" signal to shout2send
- (fixes #142954)
+ tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (get_float_mc_caps),
+ (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
+ Patch from #347221 adding a test for audioconvert
+ channel remappings.
-2004-05-21 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/kio/kioreceiver.cpp:
- * ext/kio/kioreceiver.h:
- fix sign comparison issues
+ gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
+ Original commit message from CVS:
+ * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
+ (gst_ssa_parse_parse_line):
+ Don't include the terminating NUL in the buffer size,
+ it's only there for extra paranoia (would add random
+ '*' characters at the end of each subtitle since the
+ terminator itself is not valid UTF-8 technically).
+ Also fix indenting after boilerplate macro.
-2004-05-21 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/cdxaparse/gstcdxaparse.c:
- * gst/cdxaparse/gstcdxaparse.h:
- some renaming
- add some checks/sanity
- prepare for seek addition
+ 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.
- * sys/sunaudio/gstsunaudio.c:
- remove exported dupe init function
+2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
-2004-05-21 Jan Schmidt <thaytan@mad.scientist.com>
+ * 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).
- * 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.
+2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-21 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ 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.
- * m4/as-arts.m4:
- sync with upstream version to fix test on FC2
- readd with -ko to preserve Id header
+2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-20 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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
+2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
- * sys/sunaudio/gstsunaudio.c: change category to Sink/Audio
+ 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.
- * configure.ac: enable speex plugin for speex 1.1.5+
- * ext/speex/gstspeexenc.c: fix cast warning
+2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/esd/README: fix typo
+ ext/alsa/gstalsamixertrack.c: Some const-ification.
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
+ Some const-ification.
-2004-05-20 David Schleef <ds@schleef.org>
+2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
+ Improve checking if we are dealing with a stream. Added some
+ more uris that need buffering.
-2004-05-20 David Schleef <ds@schleef.org>
+2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- * sys/Makefile.am:
+ ext/vorbis/vorbisdec.c: Remove unused variable.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
+ Remove unused variable.
-2004-05-20 David Schleef <ds@schleef.org>
+2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.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:
+ Makefile.am: include lcov.mak
+ Original commit message from CVS:
+ * Makefile.am:
+ include lcov.mak
+ * configure.ac:
+ add GCOV_LIBS to GST_LIBS
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
- * 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
+ ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
+ Original commit message from CVS:
+ Patch by: Michael Sheldon <webmaster at mikeasoft com>
+ * ext/alsa/gstalsasrc.c:
+ Add 32 bps to template caps and increase channels range
+ from [1,2] to [1,MAX]. See #346326.
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisenc.c: (vorbis_granule_time_copy),
- (gst_vorbisenc_buffer_from_packet), (gst_vorbisenc_push_buffer),
- (gst_vorbisenc_push_packet), (gst_vorbisenc_chain):
- split up push_packet into two functions
+ gst-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).
-2004-05-20 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- * 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
+ 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
-2004-05-19 Wim Taymans <wim@fluendo.com>
+2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/law/mulaw-conversion.c: (mulaw_encode):
- Fix overflow bug in ulaw encoding.
+ 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
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- don't unref the event twice
+ 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.
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- remove -Wno-sign-compare
+ 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).
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+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.
- * configure.ac:
- remove -DG_DISABLE_DEPRECATED. It's not usable without workarounds
- if you want to work against glib 2.2 and 2.4
+2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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'.
- * gst/debug/tests.c: (md5_get_value):
- fix segfault on gst-inspect
+2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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).
- * 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
+2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-19 Johan Dahlin <johan@gnome.org>
+ 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.
- * ext/dv/gstdvdec.c (gst_dvdec_quality_get_type): Add proper
- ending of the array. Fixes gst-inspect segfault on ppc.
+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).
-2004-05-19 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/dirac/gstdiracdec.cc : change category to Codec/Decoder/Video
+ ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
+ (gst_text_overlay_video_chain):
+ g_markup_escape_text() REALLY doesn't like non-UTF8 input
+ and doesn't validate its input either (and neither did
+ textoverlay it seems). Let's do that then and fix #345206.
- * m4/a52.m4 : don't fix a test that should fail with current a52dec lib
+2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
+ Original commit message from CVS:
+ * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
+ (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
+ (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
+ (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
+ (find_syncframe), (find_limits), (assign_value),
+ (count_burst_unit), (gst_multi_fd_sink_new_client),
+ (gst_multi_fd_sink_handle_client_write),
+ (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
+ (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
+ (gst_multi_fd_sink_change_state):
+ * gst/tcp/gstmultifdsink.h:
+ Added shiny new burst-on-connect methods.
+ Add properties to control the minimal amount of data queued.
+ Small cleanups.
+ API: bytes-min property
+ API: time-min property
+ API: buffers-min property
+ API: burst-unit property
+ API: burst-value property
+ API: add-full signal
+ * gst/tcp/gsttcp-marshal.list:
+ Added new marshaller code for the new signal.
+ * tests/check/elements/multifdsink.c: (GST_START_TEST),
+ (multifdsink_suite):
+ Added testcases for new burst methods.
+
+2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
-2004-05-18 David Schleef <ds@schleef.org>
+ * gst-plugins-base.spec.in:
+ update for latest changes
+ Original commit message from CVS:
+ update for latest changes
- * gst/ffmpegcolorspace/imgconvert.c: (img_convert): Fixes for
- warnings (bugs, actually) noticed by gcc but not forte.
+2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-05-18 David Schleef <ds@schleef.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
- * sys/sunaudio/Makefile.am:
- * sys/sunaudio/gstsunaudio.c: New sunaudiosink
+2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
-2004-05-18 David Schleef <ds@schleef.org>
+ 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
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
- (gst_qtdemux_loop_header): Patch from dcm@acm.org (David Moore)
- to allow qtdemux to use non-seekable streams. (bug #142272)
+2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ ChangeLog surgery
+ Original commit message from CVS:
+ ChangeLog surgery
- * gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16),
- (gst_resample_sinc_ft_float): Remove use of static temporary
- buffer. This code was obviously not supposed to last long, but
- it's stuck in our ABI, so it required a little hack to make it
- ABI-compatible. Fixes #142585.
- * gst-libs/gst/resample/resample.h: same.
+2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 David Schleef <ds@schleef.org>
+ configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
+ Original commit message from CVS:
+ * configure.ac:
+ Fix --disable-extern (can't set conditionals conditionally,
+ #343602).
- * 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
+2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 Wim Taymans <wim@fluendo.com>
+ 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.
- * 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.
+2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * 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
+2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-05-18 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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.
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- remove leftover debugging g_print
+2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-17 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (set_hwparams):
+ If we fail to set the buffer_time and period_time alsa
+ parameters, post a warning and leave alsa select a
+ default instead of failing. Fixes #342085
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- Fix for when the first format in a discont event is not a
- byte-based one. Should fix #137710.
+2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-18 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ * 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)
- * m4/a52.m4 : fix compilation with -Wall -Werror
- * m4/libfame.m4 : idem
- * m4/libmikmod.m4 : idem
+2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-17 Benjamin Otte <otte@gnome.org>
+ Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
+ out in the header file and shouldn't be listed in the docs.
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ Fix it so that it doesn't crash in the debug statement.
+
+2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ docs/libs/: add remaining symbols into correct setions
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ add remaining symbols into correct setions
+ * gst-libs/gst/audio/gstringbuffer.c:
+ fix incomplete docs
+ * gst-libs/gst/audio/gstringbuffer.h:
+ comment out not yet implemented function
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst-libs/gst/netbuffer/gstnetbuffer.c:
+ add short descriptions
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ fix return value docs
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ simplify debug logging
+ * gst-libs/gst/riff/riff-read.h:
+ sync function prototype and docs
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ remove left over symbol
+
+2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- signal the new tags before giving up the reference
+ Use GST_PLUGIN_DOCS macro in configure.ac, add
+ Original commit message from CVS:
+ * autogen.sh:
+ * configure.ac:
+ * docs/Makefile.am:
+ Use GST_PLUGIN_DOCS macro in configure.ac, add
+ --enable-plugin-docs default to autogen.sh and use
+ ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
-2004-05-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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)
+ ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
+ (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
+ (gst_ogg_demux_loop):
+ Combine GstFlowReturn from the source pads to give a
+ meaningfull result to the upstream peer or to stop the
+ processing task in case of errors.
-2004-05-17 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac : fix compilation of v4l2src with "-Wall -Werror"
- fixes #142664
+ gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c: (cb_probe):
+ Try GST_TAG_CODEC as fallback when extracting the
+ codec name; more debug info.
-2004-05-17 Benjamin Otte <otte@gnome.org>
+2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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)
+ ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
+ Original commit message from CVS:
+ * ext/ogg/Makefile.am:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
+ Extract language tags from ogm subtitle streams, so that
+ the subtitle menu choices are labelled correctly in
+ Totem (fixes #344708).
-2004-05-17 Benjamin Otte <otte@gnome.org>
+2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- use correct variable when determining amount of data to skip so we
- don't skip into the void and segfault
+ ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
+ Original commit message from CVS:
+ Patch by: Alessandro Decina <alessandro at nnva dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
+ (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
+ (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
+ (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
+ Fix various leaks. Fixes #343699.
+ Add x-smoke mime type.
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Hi, I'm a memleak
+ gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-ids.h:
+ Add IDs for 'bext' chunks (see #343837).
-2004-05-16 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
+2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-05-16 Arwed v. Merkatz <v.merkatz@gmx.net>
+ 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).
- reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst-libs/gst/audio/audioclock.c:
- Fix wrong return type (#142205).
+ po/POTFILES.in: add missing files containing translatable strings
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ add missing files containing translatable strings
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/mad/gstmad.c: (gst_mad_class_init), (gst_mad_init):
- Ignore CRCs by default (fixes #142566).
+ 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
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.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.
+ ignore more
+ Original commit message from CVS:
+ * docs/libs/.cvsignore:
+ * tests/check/elements/.cvsignore:
+ * tests/check/libs/.cvsignore:
+ ignore more
-2004-05-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.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.
+ 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)
-2004-05-16 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ 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
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_caps), (gst_alsa_get_caps):
- simplify caps
+ * 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
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
+ Original commit message from CVS:
+ * configure.ac:
+ use GST_PKG_CHECK_MODULES, cleans up output
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * common:
+ * win32/common/config.h:
+ update to cvs
+ Original commit message from CVS:
+ update to cvs
-2004-05-16 Benjamin Otte <otte@gnome.org>
+2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- add error checking to snd_pcm_delay and remove duplicate call to
- snd_pcm_delay that caused issues (see inline code comments)
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
- make more readable and fix return value when snd_pcm_delay fails
- (fixes #142586)
+ ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
+ Add support for burn:// URIs (#343385); const-ify things a bit,
+ use G_N_ELEMENTS instead of hard-coded array size.
-2004-05-15 Jan Schmidt <thaytan@mad.scientisti.com>
- * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type),
- (gst_pixbufscale_get_type), (gst_pixbufscale_base_init),
- (gst_pixbufscale_class_init), (gst_pixbufscale_getcaps),
- (gst_pixbufscale_link), (gst_pixbufscale_init),
- (gst_pixbufscale_handle_src_event), (pixbufscale_scale),
- (gst_pixbufscale_chain), (gst_pixbufscale_set_property),
- (gst_pixbufscale_get_property), (pixbufscale_init):
- * ext/gdk_pixbuf/pixbufscale.h:
- Add these files I forgot earlier
+2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-05-15 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/gdk_pixbuf/Makefile.am:
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- Add new pixbufscale element to scale RGB video
- using gdk_pixbuf, because gdk_pixbuf does BILINEAR
- and HYPER interpolation correctly.
- * ext/theora/theoraenc.c: (theora_enc_chain),
- Discard buffer and return if explicit caps could not be set
- (theora_enc_get_property):
- Make _get return kbps for the bitrate consistent with
- the _set function.
+ gst/subparse/samiparse.c: 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).
+2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-05-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * ChangeLog:
+ * configure.ac:
+ back to trunk
+ Original commit message from CVS:
+ back to trunk
- * ext/libvisual/visual.c: (gst_visual_chain):
- add missing visual_audio_analyze
+=== release 0.10.8 ===
-2004-05-14 David Schleef <ds@schleef.org>
+2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.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().
+ * 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
-2004-05-14 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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
+ 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
-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"
+2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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)
+ move last template doc snippets to source code and delete them
+ Original commit message from CVS:
+ * docs/libs/tmpl/gstaudio.sgml:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/libs/tmpl/gstmixer.sgml:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * docs/libs/tmpl/gsttuner.sgml:
+ * docs/libs/tmpl/gstxoverlay.sgml:
+ * gst-libs/gst/audio/audio.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/mixer.c:
+ * gst-libs/gst/interfaces/tuner.c:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ move last template doc snippets to source code and delete them
- * gst/cdxaparse/gstcdxaparse.c :
- * gst/cdxaparse/gstcdxaparse.h :
- - partial rewrite using RiffRead (ripped iain's wavparse code)
+2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/rtp/gstrtpL16enc.c : typo
- * gst/rtp/gstrtpgsmenc.c : typo
+ * 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.
-2004-05-13 Benjamin Otte <otte@gnome.org>
+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.
- * configure.ac:
- check for exact version of libvisual, it's not supposed to be
- API/ABI stable yet
-
-2004-05-13 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
- signal no-more-pads
-
-2004-05-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_src_convert)
- Report which format was used for GST_FORMAT_DEFAULT
- * gst/debug/Makefile.am:
- * gst/debug/gstdebug.c: (plugin_init):
- * gst/debug/progressreport.c: (gst_progressreport_base_init),
- (gst_progressreport_class_init), (gst_progressreport_init),
- (gst_progressreport_report), (gst_progressreport_set_property),
- (gst_progressreport_get_property), (gst_progressreport_chain),
- (gst_progressreport_plugin_init):
- Add progressreport element for testing.
-
-2004-05-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_change_state):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_playback_init):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_capture_init):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
- (gst_v4lsrc_grab_frame):
- add more debugging
- send a discont at start
-
-2004-05-12 Colin Walters <walters@redhat.com>
-
- * gst/asfdemux/gstasfdemux.c (gst_asf_demux_process_segment): Avoid
- inflooping if we can't find a chunk. Or in other words, don't blow
- chunks if we don't have a chunk to blow.
-
-2004-05-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/audiofile/gstafsrc.c: (gst_afsrc_get):
- Remove old debug output
- * ext/dv/gstdvdec.c: (gst_dvdec_quality_get_type),
- (gst_dvdec_class_init), (gst_dvdec_loop), (gst_dvdec_change_state),
- (gst_dvdec_set_property), (gst_dvdec_get_property):
- Change the quality setting to an enum, so it works from gst-launch
- Don't renegotiate a non-linked pad. Allows audio only decoding.
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_getcaps),
- (gst_deinterlace_link), (gst_deinterlace_init):
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
- (gst_videodrop_link):
- Some caps negotiation fixes
-
-2004-05-12 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * ext/tarkin/gsttarkin.c :
- - Change RANK from NONE to PRIMARY
- * ext/gdk_pixbuf/gstgdkpixbuf.c :
- - Change RANK from NONE to MARGINAL
- * ext/divx/gstdivxenc.c :
- - Change RANK from PRIMARY to NONE (encoder/spider issue)
-
-2004-05-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbisenc.c: (vorbis_granule_time_copy),
- (gst_vorbisenc_push_packet):
- copy a function that was added between 1.0 and 1.0.1 until we
- depend on worthwhile features of post-1.0
-
-2004-05-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac:
- enable shout2 by default
- * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
- (gst_shout2send_base_init), (gst_shout2send_init),
- (gst_shout2send_connect), (gst_shout2send_change_state):
- * ext/shout2/gstshout2.h:
- make this work again. Based on a patch by Zaheer Abbas Merali (fixes
- #142262)
- * ext/theora/theora.c: (plugin_init):
- don't set rank on encoders
+ * 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
-2004-05-11 Jeremy Simon <jesimon@libertysurf.fr>
+2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Use codec_data property instead of flag1 and flag2 for wma
+ 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.
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
- * gst/cdxaparse/gstcdxaparse.c :
- - Add mpegversion to CAPS to make it link
- - Rank is as GST_RANK_SECONDARY instead of NONE
- * gst/auparse/gstauparse.c :
- - Document all audio encoding we can encounter from Solaris 9
- headers and libsndfile information.
- - Increase max. rate from 48000 to 192000 (to match other elements)
- - Don't try to play junk data between header and samples
+ ext/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.
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
- * ext/libvisual/visual.c: (gst_visual_getcaps):
- use the right caps depending on endianness (I hope)
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_plugin_init):
- use GST_RANK_NONE for all non-decoding elements or spider gets
- mighty confused
+ gst/audioconvert/audioconvert.c: More correct float->int conversion.
+ Original commit message from CVS:
+ * gst/audioconvert/audioconvert.c: (float):
+ More correct float->int conversion.
-2004-05-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_comment):
- Fix some odd cases and fix BE metadata parsing of unicode16 text.
+ 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.
-2004-05-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+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>
- * 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.
+ * 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
-2004-05-10 David Schleef <ds@schleef.org>
+2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: Add prototype Dirac support.
- * ext/Makefile.am:
- * ext/dirac/Makefile.am:
- * ext/dirac/gstdirac.cc:
- * ext/dirac/gstdiracdec.cc:
+ 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
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+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
- * gst/auparse/gstauparse.c: (gst_auparse_class_init),
- (gst_auparse_init), (gst_auparse_chain),
- (gst_auparse_change_state):
- Hack around spider. Remove me some day please.
+2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-05-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * gst/auparse/gstauparse.c: (gst_auparse_chain):
- Fix for some uninitialized variables in previous patch, also
- makes it work. Fixes #142286 while we're at it.
+2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-11 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+ 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.
- * gst/auparse/gstauparse.c:
- fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
- only unsupported formats are ADPCM/CCITT G.72x
- reviewed by Ronald
- * gst-libs/gst/audio/audio.h: adds 24bit depth to PCM (x-raw-int)
+2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/pango/gsttextoverlay.c: Added some FIXMEs.
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
+ (gst_text_overlay_text_event):
+ Added some FIXMEs.
+
+2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/adder/gstadder.*: Implement release_request_pad.
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
+ (gst_adder_request_new_pad), (gst_adder_release_pad):
+ * gst/adder/gstadder.h:
+ Implement release_request_pad.
+ Make padcounter atomic.
+ * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
+ Added check for release_pad in adder.
+
+2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Fix build again.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
+ Fix build again.
+
+2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
+ (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
+ (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
+ (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
+ (gst_ogg_demux_bisect_forward_serialno),
+ (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
+ (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
+ add more debugging
+ clean up printf formats for granulepos and serialno
-2004-05-10 Wim Taymans <wim@fluendo.com>
+2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/vorbis/Makefile.am:
- * ext/vorbis/README:
- * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_formats),
- (oggvorbisenc_get_type), (vorbis_caps_factory), (raw_caps_factory),
- (gst_oggvorbisenc_base_init), (gst_oggvorbisenc_class_init),
- (gst_oggvorbisenc_sinkconnect), (gst_oggvorbisenc_convert_src),
- (gst_oggvorbisenc_convert_sink),
- (gst_oggvorbisenc_get_query_types), (gst_oggvorbisenc_src_query),
- (gst_oggvorbisenc_init), (gst_oggvorbisenc_get_tag_value),
- (gst_oggvorbisenc_metadata_set1), (gst_oggvorbisenc_set_metadata),
- (get_constraints_string), (update_start_message),
- (gst_oggvorbisenc_setup), (gst_oggvorbisenc_write_page),
- (gst_oggvorbisenc_chain), (gst_oggvorbisenc_get_property),
- (gst_oggvorbisenc_set_property), (gst_oggvorbisenc_change_state):
- * ext/vorbis/oggvorbisenc.h:
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisenc.c: (vorbis_caps_factory),
- (raw_caps_factory), (gst_vorbisenc_class_init),
- (gst_vorbisenc_init), (gst_vorbisenc_setup),
- (gst_vorbisenc_push_packet), (gst_vorbisenc_chain),
- (gst_vorbisenc_get_property), (gst_vorbisenc_set_property):
- * ext/vorbis/vorbisenc.h:
- Added a raw vorbis encoder to be used with the oggmuxer.
- We still need the old encoder for some gnome applications,
- read the README to find out how that works.
- The raw encoder is called "rawvorbisenc" until 0.9.
-
-2004-05-10 Wim Taymans <wim@fluendo.com>
-
- * ext/ogg/gstogg.c: (plugin_init):
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init),
- (gst_ogg_print):
- * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type),
- (gst_ogg_mux_base_init), (gst_ogg_mux_class_init),
- (gst_ogg_mux_get_sink_event_masks), (gst_ogg_mux_init),
- (gst_ogg_mux_sinkconnect), (gst_ogg_mux_pad_link),
- (gst_ogg_mux_pad_unlink), (gst_ogg_mux_request_new_pad),
- (gst_ogg_mux_handle_src_event), (gst_ogg_mux_next_buffer),
- (gst_ogg_mux_push_page), (gst_ogg_mux_compare_pads),
- (gst_ogg_mux_queue_pads), (gst_ogg_mux_loop),
- (gst_ogg_mux_get_property), (gst_ogg_mux_set_property),
- (gst_ogg_mux_change_state), (gst_ogg_mux_plugin_init):
- Added an ogg muxer.
- Small typo fixes in the demuxer.
-
-2004-05-10 Wim Taymans <wim@fluendo.com>
-
- * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
- (theora_enc_sink_link), (theora_push_packet), (theora_enc_chain),
- (theora_enc_change_state), (theora_enc_set_property),
- (theora_enc_get_property):
- Mark the last packet with an EOS flag which is not really needed
- in gstreamer.
- Do some better video framerate initialisation.
- Update the buffer timestamp.
-
-2004-05-10 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_change_state):
- Return the result of the parent state change call
-
-2004-05-10 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
-
- * gst/law/alaw.c : alawdec should be registered with type ALAWDEC, not ALAWENC
- * gst/law/alaw-decode.c : put audio/x-alaw on pads, instead of audio/x-mulaw
- * gst/law/alaw-encode.c : (idem)
- * ext/a52dec/gsta52dec.c : mark audio/a52, audio/ac3 as deprecated in a comment
- * gst/ac3parse/gstac3parse.c : audio/ac3 => audio/x-ac3
- * gst/realmedia/rmdemux.c : audio/a52 => audio/x-ac3
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- don't use a fixed buffer size when writing variable length data to
- it. Fixes memory corruption and makes alsasrc work
-
-2004-05-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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.
- * ext/gnomevfs/gstgnomevfssink.c:
- (_gst_boolean_allow_overwrite_accumulator),
- (gst_gnomevfssink_class_init), (gst_gnomevfssink_open_file):
- Run glib's default signal handler (??) in RUN_CLEANUP rather than
- RUN_LAST, and don't use that to set the accumulator value because
- then it's always FALSE.
+2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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.
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data),
- (gst_riff_create_audio_caps),
- (gst_riff_create_audio_template_caps):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
- Fix for unaligned RIFF files (i.e. where all the chunks together
- in a LIST chunk are not of the same size as the size given in
- the LIST chunk header). Fixes several odd WAVE files. Also fix
- ADPCM (block_align property) in audio, so that wavparse based
- on this works now as it used to stand-alone.
-
-2004-05-09 Edward Hervey <bilboed@bilboed.com>
-
- reviewed by Benjamin Otte <otte@gnome.org>
-
- * ext/a52dec/gsta52dec.c:
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
- * ext/faac/gstfaac.c: (gst_faac_base_init):
- * ext/faad/gstfaad.c: (gst_faad_base_init):
- * ext/ivorbis/vorbisfile.c:
- * ext/lame/gstlame.c:
- * ext/libfame/gstlibfame.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
- * ext/sidplay/gstsiddec.cc:
- * ext/speex/gstspeexdec.c:
- * ext/speex/gstspeexenc.c:
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- correct klasses. Mostly s,Codec/(Audio|Video),\1/Codec,
- (fixes #142193)
-
-2004-05-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsa.c: (device_list),
- (gst_alsa_class_probe_devices):
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
- Fix alsa oddness in mixer after the combination of using mixer
- in source/sink elements and using hw:x,y instead of just hw:x.
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * gst/wavparse/gstwavparse.c: (gst_wavparse_destroy_sourcepad),
- (gst_wavparse_create_sourcepad):
- make PAUSED=>READY=>PAUSED=READY work by not destroying NULL
- sourcepads
-
-2004-05-09 Benjamin Otte <otte@gnome.org>
-
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- allow discont events before caps nego
-
-2004-05-08 Benjamin Otte <otte@gnome.org>
-
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- don't leak events
-
-2004-05-08 Benjamin Otte <otte@gnome.org>
-
- * gst/level/gstlevel.c: (gst_level_link), (gst_level_chain),
- (gst_level_change_state), (gst_level_init):
- * gst/level/gstlevel.h:
- figure out if we're initialized directly instead of keeping a
- variable that's wrong in 90% of cases
- don't initialize pads and then leak them and use a new unitialized
- pad. (fixes #142084)
- these were bugs so n00bish I didn't find them for an hour :/
-
-2004-05-08 Iain <iain@prettypeople.org>
-
- * gst/wavparse/gstwavparse.[ch]: Rewrote to use RiffRead instead.
- * gst-libs/gst/riff/riff-read.c (gst_riff_read_peek_head): Unstatic it
- (gst_riff_read_element_data): Ditto, and added a got_bytes argument to
- return the length that was read.
- (gst_riff_read_strf_auds): Allow fmt tags as well.
-
-2004-05-07 David Schleef <ds@schleef.org>
-
- * ext/faad/gstfaad.c: (gst_faad_sinkconnect): HACK to correct
- signed char assumption in faad.h.
-
-2004-05-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps):
- Missing break, detected by Daniel Gazard <daniel.gazard@free.fr>.
-
-2004-05-07 Colin Walters <walters@redhat.com>
-
- * gst/volume/gstvolume.c (gst_volume_dispose): Unref dpman.
- * ext/flac/gstflacdec.c (gst_flacdec_dispose): Add dispose
- function.
- * gst/audioscale/gstaudioscale.c (gst_audioscale_dispose):
- Add dispose function.
-
-2004-05-08 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/dv/gstdvdec.c: (gst_dvdec_video_link):
- Fix caps nego and pad templates. RGB mode caps should
- work now.
- * ext/dvdnav/gst-dvd:
- Move mpeg2dec inside the thread because otherwise the
- queue rejects cap changes mid-stream
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_get_type),
- (gst_mpeg2dec_flush_decoder):
- For mpeg2dec > 0.4.0, call the flush function instead of
- manually extracting all in-flight frames.
- * ext/raw1394/gstdv1394src.c: (gst_dv1394src_factory),
- (gst_dv1394src_init), (gst_dv1394src_iso_receive):
- Change mime type video/dv go video/x-dv to match the
- rest of gst-plugins
-
-2004-05-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_build_list):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_get_type),
- (gst_alsa_sink_class_init):
- * ext/alsa/gstalsasink.h:
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_get_type),
- (gst_alsa_src_class_init):
- * ext/alsa/gstalsasrc.h:
- Make alsasink/src a subclass of alsamixer so that mixer stuff
- shows up in gst-rec. Needs some finetuning.
+2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ 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.
- * ext/lame/gstlame.c: (gst_lame_chain):
- simplify
- * ext/mad/gstmad.c: (gst_mad_handle_event):
- fix event leak
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
- be able to detect mp3 files < 4096 bytes
+2006-05-29 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).
-2004-05-06 Wim Taymans <wim@fluendo.com>
+2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
- * 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.
+ 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.
-2004-05-06 Wim Taymans <wim@fluendo.com>
+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.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Forward events first before deciding that negotiation was
- not performed.
+2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- First process the events before deciding that negotiation
- was not performed.
+2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-05-06 Wim Taymans <wim@fluendo.com>
+ 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.
- * ext/theora/Makefile.am:
- * ext/theora/theora.c: (plugin_init):
- * ext/theora/theoradec.c: (theora_dec_change_state):
- * ext/theora/theoraenc.c: (gst_theora_enc_base_init),
- (gst_theora_enc_class_init), (gst_theora_enc_init),
- (theora_enc_sink_link), (theora_enc_event), (theora_push_packet),
- (theora_enc_chain), (theora_enc_change_state),
- (theora_enc_set_property), (theora_enc_get_property):
- Added a theora encoder, grouped the encoder and decoder into the
- same plugin.
-
-2004-05-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
- (gst_jpegenc_chain):
- fix DURATION on outgoing buffers
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
- debug using time formats
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_sink_link):
- windows with width/height 0 generate X errors, so don't allow them
-
-2004-05-05 Wim Taymans <wim@fluendo.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c: (src_templ),
- (gst_mpeg2dec_base_init), (gst_mpeg2dec_init),
- (gst_mpeg2dec_negotiate_format):
- * ext/mpeg2dec/gstmpeg2dec.h:
- removed the static pad template so that we can add the
- more accurate framerate value to the caps.
-
-
-2004-05-04 Benjamin Otte <otte@gnome.org>
+2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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)
+ * ChangeLog:
+ Mention bugs fixed by previous commit
+ Original commit message from CVS:
+ Mention bugs fixed by previous commit
+
+2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
+ (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
+ (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
+ (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
+ Improve the errors produced on bad output, including some human
+ readable description strings.
+ Handle RGB Xv formats properly by transforming them into our
+ big-endian caps description.
+ Use gst_caps_truncate to ensure that we never try and choose a
+ non-fixed caps in buffer_alloc.
+ Handle the case where the XServer has a different idea about the size
+ required for a particular frame and gives us too small a memory
+ allocation.
+ Use -1 to indicate 'no image format', because 0 is a valid XServer
+ image format number.
+ Put RGB Xv formats at the end of the caps, so that we always prefer
+ YUV format frames.
+ Iterate the available Xv Encodings to determine the maximum width and
+ height, and then return that in our caps.
+
+2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
+ When there is only one unfinished pad and it receives an event that
+ doesn't match our requirements, we need to set alldone=FALSE so that
+ the fakesink is not removed yet.
+
+2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
+ Use gst_type_find_helper_for_buffer() to find the type
+ of stream from the first packet.
+ * configure.ac:
+ Bump requirements to core CVS (needed for vorbis
+ typefinding to work).
-2004-05-04 Wim Taymans <wim@fluendo.com>
+2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/mpeg2enc/gstmpeg2enc.cc: (gst_mpeg2enc_dispose): call the
- parent dispose function to avoid segfault on destroy.
+ gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
+ Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
+ Else they play perfectly fine with qtdemux.
-2004-05-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ make more debug catagories static
+ Original commit message from CVS:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
+ * gst/audiorate/gstaudiorate.c:
+ make more debug catagories static
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (message_received),
+ (test_event_message_received), (GST_START_TEST),
+ (test_play_twice_message_received), (adder_suite):
+ added test case for using element twice, extra bonus points for anyone
+ who can make these test run reliably
-2004-05-04 Benjamin Otte <otte@gnome.org>
+2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_dec_chain):
+ Make work with time-stamped input buffers that do not
+ have a granulepos in BUFFER_OFFSET_END (like theora
+ buffers coming from matroskademux). Fixes #342448.
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
+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.
- * Makefile.am:
- * po/af.po:
- * po/az.po:
- * po/en_GB.po:
- * po/nl.po:
- * po/sr.po:
- * po/sv.po:
- Updated translations
+2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
+ Original commit message from CVS:
+ Patch by: Peter Kjellerstedt <pkj at axis com>
+ * gst/tcp/Makefile.am:
+ fdstresstest doesn't need Gtk+, fix compilation if
+ gtk is not available (#342566).
- * gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
- refactor/comment code
+2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-05-02 Ronald 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()
- * gst/asfdemux/Makefile.am:
- * gst/asfdemux/asfheaders.c:
- * gst/asfdemux/asfheaders.h:
- * gst/asfdemux/gstasf.c: (plugin_init):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_get_type),
- (gst_asf_demux_base_init), (gst_asf_demux_process_comment),
- (gst_asf_demux_setup_pad):
- * gst/asfdemux/gstasfdemux.h:
- * gst/asfdemux/gstasfmux.c:
- * gst/asfdemux/gstasfmux.h:
- Add tagging support to demuxer, split out registration in its own
- file instead of in demux (hacky), and prevent having some tables
- in our memory multiple times (in asfheaders.h).
+2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-05-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ On second thought, just skip JUNK chunks automatically, so
+ the caller doesn't have to handle this. Fixes #342345.
+ Also, return GST_FLOW_UNEXPECTED if we get a short read,
+ not GST_FLOW_ERROR.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_metadata):
- * gst/matroska/matroska-ids.h:
- Basic tag reading support.
+2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
+ Don't bail out on JUNK chunks with a size of 0 (would try to
+ pull_range 0 bytes before, which sources don't like too much).
+ See #342345.
- * 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).
+2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
+ Use the gstutil scaling function to preserve 64 bits while calculating
+ output width and height from the display-aspect-ratio. (A continuation
+ of #341542)
+
+2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
+ Original commit message from CVS:
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
+ (gst_xvimagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.h:
+ When performing buffer allocations, remember the caps and image format
+ we return so that if the same caps are asked for next time we can
+ return them immediately without doing any caps intersections.
+
+2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+
+ gst-libs/gst/rtp/README: Some new documentation
+ Original commit message from CVS:
+ 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ * gst-libs/gst/rtp/README:
+ Some new documentation
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ New RTP audio base payloader class. Supports frame or sample based codecs.
+ Not enabled in Makefile.am until approved.
+
+2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
+ Original commit message from CVS:
+ * tests/check/elements/alsa.c: (test_device_property_probe):
+ Fix test case: don't try to free NULL GValueArray when there
+ are no devices.
-2004-04-30 David Schleef <ds@schleef.org>
+2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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:
+ tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ * tests/check/elements/alsa.c: (test_device_property_probe),
+ (alsa_suite), (main):
+ Add simple test that runs a device property probe on alsasrc,
+ alsasink and alsamixer. Disable valgrind check for now (too
+ many leaks in libasound, and valgrind ignored my suppressions
+ additions).
-2004-04-30 Colin Walters <walters@verbum.org>
+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>
- * ext/alsa/gstalsasink.c (gst_alsa_sink_mmap): Plug a memleak.
+2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_getcaps):
- remove broken nego fix
+2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
-2004-04-30 Benjamin Otte <otte@gnome.org>
+ 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).
- * 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 :)
+2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-29 David Schleef <ds@schleef.org>
+ 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/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.
+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.
-2004-04-29 Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+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.
- * ext/alsa/gstalsa.c : (gst_alsa_class_probe_devices)
- * ext/alsa/gstalsa.h :
- change alsa pcm device discovery to find more than 1 device
- per card. code review by Ronald.
+2006-05-15 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.
-2004-04-29 David Schleef <ds@schleef.org>
+2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/oss/gstosselement.c: (gst_osselement_rate_probe_check):
- Add a check for a driver bug on FreeBSD. (bug #140565)
+ 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.
-2004-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ 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
-2004-04-28 David Schleef <ds@schleef.org>
+2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/speed/gstspeed.c: (speed_parse_caps): Fix caps parsing.
- (bug #140064)
+ 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).
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
- Don't probe for playback device if we're a source element. Fixes
- #139658.
+ 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).
-2004-04-29 Benjamin Otte <otte@gnome.org>
+2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_chain):
- rewrite buffer offset
+ 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).
-2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.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>
+ 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).
- * 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
+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).
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * win32/common/config.h:
+ update config.h
+ Original commit message from CVS:
+ update config.h
-2004-04-27 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * ext/ogg/README:
+ fix mistakes in README
+ Original commit message from CVS:
+ fix mistakes in README
- 2004-04-26 Benjamin Otte <otte@gnome.org>
+2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.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
+ 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.
-2004-04-26 Benjamin Otte <otte@gnome.org>
+2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
* 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>
+ Back to CVS
+ Original commit message from CVS:
+ Back to CVS
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- audio/x-raw-int with height rules! not. Now it's depth.
+=== release 0.10.7 ===
-2004-04-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ 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"
-2004-04-22 Zaheer Abbas Merali <zaheerabbas at merali dot org>
+2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- reviewed by Benjamin Otte <otte@gnome.org>
+ * 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
- * ext/shout/gstshout.c: (gst_icecastsend_change_state):
- fix shoutcast not working (fixes #140844)
+2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-22 Benjamin Otte <otte@gnome.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.
- * ext/hermes/gsthermescolorspace.c:
- (gst_hermes_colorspace_caps_remove_format_info):
- * gst/colorspace/gstcolorspace.c:
- (gst_colorspace_caps_remove_format_info):
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_caps_remove_format_info):
- s/gst_caps_simplify/gst_caps_do_simplify/
+2006-05-12 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)
-2004-04-22 Benjamin Otte <otte@gnome.org>
+2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ 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).
-2004-04-20 David Schleef <ds@schleef.org>
+2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
- * ext/sdl/Makefile.am: Link against libgstinterfaces, not
- libgstxoverlay. jmmv@menta.net (Julio M. Merino Vidal)
- (bug #140384)
+ win32/MANIFEST: update win32 files listing
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ update win32 files listing
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
+2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- reviewed by David Schleef
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * 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).
+2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 David Schleef <ds@schleef.org>
+ * tests/check/elements/multifdsink.c:
+ disable failing check on gentoo64
+ Original commit message from CVS:
+ disable failing check on gentoo64
- * gst/realmedia/rmdemux.c: This was supposed to part of the
- last checkin. Same idea.
+2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Daniel Gazard <daniel.gazard@epita.fr>
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
- reviewed by David Schleef
+2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: bump required gstreamer version to 0.8.1.1
- because of following changes [--ds]
+ * tests/check/elements/multifdsink.c:
+ macros show the correct line
+ Original commit message from CVS:
+ macros show the correct line
- * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h.
- (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list)
- (gst_riff_read_header): Use GST_READ_UINT*
- macros to access possibly unaligned memory.
+2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
- * 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/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
- * gst/tags/gstvorbistag.c: (ADVANCE): Likewise
+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/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc).
- (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET)
- (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET)
- (gst_qtdemux_loop_header, gst_qtdemux_loop_header)
- (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type)
- (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT*
- macros to access possibly unaligned memory.
+2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk):
- Likewise.
+ 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).
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead)
- (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise.
+2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Likewise.
+ * gst/tcp/gstmultifdsink.c:
+ more commenting
+ Original commit message from CVS:
+ more commenting
- * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA)
- (gst_mpeg2subt_chain_subtitle): Likewise.
+2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq)
- (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain):
- Likewise.
+ 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
- * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info):
- Likewise.
+2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
- * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes):
- Likewise.
+ 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.
- * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint):
- Likewise.
+2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot 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.
- * configure.ac:
- update required version of GStreamer because of GST_TIME_FORMAT
+2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * tests/check/elements/multifdsink.c:
+ set caps and plug leaks
+ Original commit message from CVS:
+ set caps and plug leaks
- * ext/mad/gstid3tag.c: (gst_id3_tag_init):
- remove leftover g_print
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- don't try setting only a subset of the caps. We don't want to kill
- autoplugging on purpose
+2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot 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
- * sys/ximage/ximagesink.c: (plugin_init):
- * sys/xvimage/xvimagesink.c: (plugin_init):
- add debugging categories
+2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-20 Thomas Vander Stichele <thomas at apestaart dot 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
- * 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
+2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/tcp/gstmultifdsink.c:
+ DEBUG_FUNCPTR'ing
+ Original commit message from CVS:
+ DEBUG_FUNCPTR'ing
- * po/LINGUAS:
- * po/uk.po:
- Added Ukrainian translation (Maxim V. Dziumanenko)
+2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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>
- * 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.
+ 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.
-2004-04-18 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.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.
+ 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.
-2004-04-18 Benjamin Otte <otte@gnome.org>
+2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
- * 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
+ 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).
-2004-04-17 Iain <iain@prettypeople.org>
+2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/wavparse/gstwavparse.c (gst_wavparse_loop): Allow all events,
- not just EOS.
+ 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
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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
+ gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_request_new_pad),
+ (gst_adder_collected):
+ * gst/adder/gstadder.h:
+ Remove bogus segment merging and forwarding, we don't
+ care about timestamps anyway and we just produce a
+ continuous stream.
+ Also create a nice NEWSEGMENT event when we start.
+ Use _scale_int some more.
-2004-04-17 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
- * configure.ac:
- require mpeg2dec >= 0.4.0
+ tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/icles/stress-xoverlay.c:
+ Fix if core was built without parsing support.
-2004-04-17 Benjamin Otte <otte@gnome.org>
+2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_xwindow_id):
- call GST_ELEMENT_ERROR whenever get_xcontext fails. Includes
- assorted cleanup fixes.
+ gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add SEDG (Samsung MPEG-4) fourcc.
-2004-04-16 David Schleef <ds@schleef.org>
+2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
- * sys/ximage/ximagesink.h: Compile fix for FreeBSD. (bug #140268)
- * sys/xvimage/xvimagesink.h: same
+ tests/examples/volume/volume.c: Fox if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/examples/volume/volume.c:
+ Fox if core was built without parsing support.
+ * tests/examples/seek/seek.c:
+ Disable the parse_launch example if core was built without parsing
+ support.
-2004-04-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Fix GST_ELEMENT_ERROR with (NULL)
+ tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c:
+ Disable the parse_launch example if core was built without parsing
+ support.
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/riff/riff-media.c:
- (gst_riff_create_video_caps_with_data):
- Add div[3456] as fourccs for DivX 3 (fixes #140137).
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/videoscale/gstvideoscale.c:
+ doc reparagraphing and DEBUG_FUNCPTRing
+ Original commit message from CVS:
+ doc reparagraphing and DEBUG_FUNCPTRing
-2004-04-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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.
+ autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
+ Original commit message from CVS:
+ * autogen.sh: (CONFIGURE_DEF_OPT):
+ libtoolize on Darwin/MacOSX is called glibtoolize
-2004-04-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * configure.ac: bump nano to 1
+ 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
-=== release 0.8.1 ===
+2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-15 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
- * configure.ac: releasing 0.8.1, "Comforting Sounds"
+2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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
+2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
+ Original commit message from CVS:
+ * gst/adder/gstadder.c: (gst_adder_sink_event),
+ (gst_adder_request_new_pad), (gst_adder_change_state):
+ * gst/adder/gstadder.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
+ (adder_suite), (main):
+ Add sink-event handling to adder. It tries to merge incomming
+ newsegment-events. Added test to check if segment_done is comming
+ through.
- * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_plugin_init):
- lower rank of dvddemux so that it's not used for mpeg playback.
+2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
-2004-04-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * 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.
- * configure.ac:
- save libs correctly when checking mad
+2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/mad/gstid3tag.c: (plugin_init):
- lower rank of id3tag as proposed by Benjamin. Fixes #139926.
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ some RTP debug
+ Original commit message from CVS:
+ some RTP debug
-2004-04-13 David Schleef <ds@schleef.org>
+2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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.
+ gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
+ (plugin_init):
+ Refine musepack typefinding a bit. Return MAXIMUM
+ probability when we detect stream version 7 to make
+ sure the mpeg audio typefinder doesn't trump us.
-2004-04-13 David Schleef <ds@schleef.org>
+2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/avi/gstavidemux.c: (gst_avi_demux_reset): Fix memleak.
- patch from Sebastien Cote (bug #139958)
+ gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Protect against unexpected NULL strf_data buffer.
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * examples/gstplay/Makefile.am:
- * examples/gstplay/player.c: (main):
- make the commandline player example use gconf settings
+ tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
+ Original commit message from CVS:
+ * tests/check/elements/audioconvert.c: (verify_convert),
+ (GST_START_TEST):
+ interpret the out[] buffer in the order the bytes are actually
+ put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
+ Other tests should use BYTE_ORDER since the array is filled in
+ with actual values
-2004-04-13 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.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.
+ * tests/check/elements/audioconvert.c:
+ dump expected data when audioconvert test fails
+ Original commit message from CVS:
+ dump expected data when audioconvert test fails
+2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-12 Christian Schaller <Uraeus@gnome.org>
+ 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
- * sys/oss/gstosselement.c: s/lstat/stat/ from freeBSD, since it can
- be a symlink
+2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ext/ogg/gstoggmux.c:
+ * ext/theora/theoraenc.c:
+ add another include
+ Original commit message from CVS:
+ add another include
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_data):
- Handle JUNK chunks inside data section. Prevents warnings.
+2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * gst/subparse/gstssaparse.c:
+ atoi() needs stdlib.h
+ Original commit message from CVS:
+ atoi() needs stdlib.h
- * 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.
+2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
- Fix another codecname mismatch.
+2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-11 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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>
- * 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.
+2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * common:
+ * docs/Makefile.am:
+ * docs/libs/Makefile.am:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/Makefile.am:
+ * docs/upload.mak:
+ use common upload.mak
+ Original commit message from CVS:
+ use common upload.mak
- * 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.
+2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * 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).
+2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
-2004-04-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file):
- Bail out if no filename was given.
+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.
-2004-04-10 Ronald 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.
- * 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).
+2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-09 Benjamin Otte <otte@gnome.org>
+ 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
- * 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
+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>
-2004-04-08 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/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_register): Change rank to PRIMARY.
+2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-08 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/colorspace/gstcolorspace.c: Don't advertise a conversion
- we don't support (bug #139532)
+2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ChangeLog:
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * 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
+=== release 0.10.6 ===
-2004-04-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * 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
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/gconf/test-gconf.c: (main):
- add missing gst_init
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ dist more win32 files
+ Original commit message from CVS:
+ dist more win32 files
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
- free the mutexes, too
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-04-07 Benjamin Otte <otte@gnome.org>
+2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_dispose):
- actually free the URI string
- * ext/mad/gstid3tag.c: (gst_id3_tag_src_event):
- compute offset correctly when passing discont events
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event):
- don't leak discont events
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
- add some missing breaks so caps aren't copied randomly
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream):
- if we realloc memory, we better use it
+ gst/videoscale/gstvideoscale.c: Add call to oil_init().
+ Original commit message from CVS:
+ * gst/videoscale/gstvideoscale.c: Add call to oil_init().
+ Fixes #338897.
-2004-04-06 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/mad/gstmad.c: (normal_seek):
- fix GST_FORMAT_TIME usage
+ * ChangeLog:
+ * configure.ac:
+ * win32/common/config.h:
+ new prerelease
+ Original commit message from CVS:
+ new prerelease
-2004-04-05 David Schleef <ds@schleef.org>
+2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/kio/kiosrc.cpp: Undefine KDE_DEPRECATED so we can use
- a deprecated function (hack!)
+ 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.
-2004-04-05 Benjamin Otte <otte@gnome.org>
+2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
- * ext/esd/esdmon.c: (gst_esdmon_get):
- fix nonterminated vararg and memleak
+ 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.
-2004-04-05 Benjamin Otte <otte@gnome.org>
+2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.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
+ * configure.ac:
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
-2004-04-05 Stefan Kost <kost@imn.htwk-leipzig.de>
+2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- reviewed by Benjamin Otte <otte@gnome.org>
+ 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.
- * ext/ladspa/gstladspa.c: (gst_ladspa_class_init):
- check for broken LADSPA parameters (fixes #138635)
+2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-04-05 Benjamin Otte <otte@gnome.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.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps):
- advertise buffer-frames correctly on sinkpads
+2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * 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
+2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-04-05 Thomas Vander Stichele <thomas at apestaart dot 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
- * ext/mad/gstmad.c: add debugging category, comment + cleanups
+2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.net>
+ 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
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
* configure.ac:
- fix == in test(1) operator
-
-2004-04-05 Julio M. Merino Vidal <jmmv@menta.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 libtool versioning
+ Original commit message from CVS:
+ update libtool versioning
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
* configure.ac:
- fix --export-symblos-regex to a working regex.
+ * win32/common/config.h:
+ prerelease
+ Original commit message from CVS:
+ prerelease
-2004-04-04 Benjamin Otte <otte@gnome.org>
+2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
- * sys/oss/.cvsignore:
- add for oss_probe
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
+ Original commit message from CVS:
+ Patch by: Antoine Tremblay <hexa00 at gmail dot com>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
+ Fix some memory leaks: on finalize, free buffers left in the queue
+ before destroying the queue; in _push(), unref rtp_buf even if
+ the process vfunc returned a NULL buffer as output buffer (#337548);
+ demote some recuring debug messages to LOG level.
-2004-04-03 Tim-Phillip Müller <t.i.m@zen.co.uk>
+2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
- reviewed by Benjamin Otte <otte@gnome.org>
+ * 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.
- * 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)
+2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-04-02 David Schleef <ds@schleef.org>
+ * ChangeLog:
+ Mention bug #336617 closed by recent commit
+ Original commit message from CVS:
+ Mention bug #336617 closed by recent commit
- * ext/gdk_pixbuf/Makefile.am: Make sure gstgdkanimation.h is
- disted (bug #138914)
+2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
-2004-04-01 Benjamin Otte <otte@gnome.org>
+ 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.
- * 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.
+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
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst-libs/gst/riff/riff-media.c:
- fail on error, don't try to set stuff on NULL caps
+ 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.
-2004-03-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
- * 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:
+ 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
-2004-03-28 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)
+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.
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Fix buffer overflow read error.
+ 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.
-2004-03-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
- * 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...
+ * 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.
-2004-03-23 Jeremy Simon <jesimon@libertysurf.fr>
+2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
- * gst/typefind/gsttypefindfunctions.c: (ape_type_find),
- (plugin_init):
- Add a monkeysaudio typefind function
+ 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!
-2004-03-23 Johan Dahlin <johan@gnome.org>
+2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
- * 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.
+ 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.
-2004-03-22 David Schleef <ds@schleef.org>
+2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
- * sys/oss/Makefile.am:
- * sys/oss/oss_probe.c: (main), (probe_check), (add_range),
- (check_rate), (add_rate): Rate probing test app.
+ ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
+ Original commit message from CVS:
+ 2006-04-04 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoradec.c (theora_handle_data_packet): This value
+ is often negative -- make it signed so as not to wrap around.
+ Fixes segfaults introduced on 9 March.
-2004-03-21 Benjamin Otte <otte@gnome.org>
+2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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)
+ ext/theora/: Don't try to store a gdouble in a gboolean.
+ Original commit message from CVS:
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (theora_dec_src_event):
+ Don't try to store a gdouble in a gboolean.
+ Small cleanups.
-2004-03-20 David I. Lehn <dlehn@users.sourceforge.net>
+2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac: GST_PACKAGE default: s/GStreamer/GStreamer Plugins/
+ ext/ogg/gstoggmux.c: Oggmux sucks.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
+ Oggmux sucks.
+ Make it suck slightly less by writing out the final page.
+ Still can't encode a vorbis-in-ogg file correctly, though.
-2004-03-20 Tim-Phillip Müller <t.i.m@zen.co.uk>
+2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
- reviewed by: Benjamin Otte <otte@gnome.org>
+ 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.
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- terminate gst_event_new_discontinuous correctly (fixes parts of
- #137711)
+2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
-2004-03-19 David Schleef <ds@schleef.org>
+ 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.
- * gst-libs/gst/Makefile.am: Enable xoverlay unconditionally,
- since it doesn't depend on X, and it's part of our ABI.
+2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
-2004-03-19 Iain <iain@prettypeople.org>
+ 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.
- * gst/interleave/deinterleave.c (deinterleave_sink_link): Use the
- is_int in the structure, not the local variable.
+2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-19 David Schleef <ds@schleef.org>
+ * 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
- * 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.
+2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ * common:
+ * ext/pango/gsttextoverlay.c:
+ * sys/v4l/gstv4lsrc.c:
+ remove BT8x8 from description, works for more devices
+ Original commit message from CVS:
+ remove BT8x8 from description, works for more devices
- * po/LINGUAS:
- * po/af.po:
- adding Afrikaans (Petri Jooste)
+2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ Fixed the sample pipeline (see #323798)
- * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
- (gst_ffmpegcolorspace_chain):
- throw error instead of g_critical (#137588)
+2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
+ use AS_VERSION and AS_NANO more cleanups
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ * win32/common/config.h.in:
+ use AS_VERSION and AS_NANO
+ more cleanups
- * Makefile.am:
- * configure.ac:
- dist common and m4 correctly
- * po/sv.po:
+2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
+ uninitialized variable return that would happen.
+
+2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
+ uninitialized variable return that would never happen.
+
+2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ Original commit message from CVS:
+ 2006-03-31 Andy Wingo <wingo@pobox.com>
+ * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
+ (vorbis_parse_sink_event): Add an event function to flush our
+ state on a seek, and to drain buffers on a premature EOS.
+ (vorbis_parse_push_headers, vorbis_parse_clear_queue)
+ (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
+ (vorbis_parse_chain, vorbis_parse_queue_buffer)
+ (vorbis_parse_drain_queue): Queue up buffers until we can set
+ their timestamps and granulepos values.
+ * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
+ and keep track of data needed for deriving granulepos and
+ timestamps for buffers.
+
+2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-17 David Schleef <ds@schleef.org>
+ * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
+ * pkgconfig/gstreamer-plugins-base.pc.in:
+ expose pluginsdir so gonlin can use it for tests
+ Original commit message from CVS:
+ expose pluginsdir so gonlin can use it for tests
- * pkgconfig/gstreamer-media-info.pc.in: Add Version.
- (bug #137348)
+2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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
- * po/LINGUAS:
- * po/sv.po:
- adding Swedish translation (Christian Rose)
+2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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
- * Makefile.am: use release.mak
+2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * common/ChangeLog:
- * common/gst-autogen.sh:
- add some explanation about the version detection
- * configure.ac:
- fix X check
+2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * configure.ac: bump nano to 1
+2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.8.0 ===
+ gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Don't post RESOURCE_NOT_FOUND error when we can't find a source
+ element for a particular protocol, that's confusing for users.
+ Instead, post a RESOURCE_FAILED error, so that our own error
+ message is actually shown in totem etc. (#336303).
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
- * configure.ac: release 0.8.0, "Pharmaceutical Itch"
+ ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
+ Original commit message from CVS:
+ Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_get_icy_metadata):
+ Fix some minor memory leaks (#336194).
-2004-03-16 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- update libtool version
- * gst-libs/gst/media-info/Makefile.am:
- actually use libtool version
+ ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfs.c:
+ (gst_gnome_vfs_location_to_uri_string):
+ * ext/gnomevfs/gstgnomevfs.h:
+ * ext/gnomevfs/gstgnomevfssink.c:
+ (gst_gnome_vfs_sink_set_property):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
+ Make gnomevfssink accept filenames as well as URIs for the
+ "location" property, just like gnomevfssrc does (and
+ filesrc/filesink do) (#336190).
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: fix speex detection to work with 1.0 but not 1.1
+ tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
+ Original commit message from CVS:
+ * tests/check/generic/clock-selection.c: (GST_START_TEST):
+ set to NULL before unreffing, fixes a valgrind leak.
+ Why was this not triggering the error that an object needs to
+ be NULL before unreffing ?
+ * win32/common/config.h:
+ update
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gst-plugins.spec.in:
- * pkgconfig/Makefile.am:
- * pkgconfig/gstreamer-gconf-uninstalled.pc.in:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs.pc.in:
- * pkgconfig/gstreamer-media-info-uninstalled.pc.in:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins-uninstalled.pc.in:
- * pkgconfig/gstreamer-plugins.pc.in:
- remove @VERSION@ from some of the pc files since core and plugins
- are decoupled.
- created gstreamer-plugins.pc as it's a better name, but keeping
- -libs around for now to get fixes upstream done first.
+ gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (convert_encoding),
+ (gst_sub_parse_change_state):
+ * gst/subparse/gstsubparse.h:
+ Text subtitle files may or may not be UTF-8. If it's not, we
+ don't really want to see '?' characters in place of non-ASCII
+ characters like accented characters. So let's assume the input
+ is UTF-8 until we come across text that is clearly not. If it's
+ not UTF-8, we don't really know what it is, so try the following:
+ (a) see whether the GST_SUBTITLE_ENCODING environment variable
+ is set; if not, check (b) if the current locale encoding is
+ non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
+ the current locale encoding is UTF-8 and the environment variable
+ was not set to any particular encoding. Not perfect, but better
+ than nothing (and better than before, I think) (fixes #172848).
-2004-03-15 Julien MOUTTE <julien@moutte.net>
+2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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:
+ * docs/plugins/tmpl/.gitignore:
+ * tests/check/libs/.gitignore:
+ * tests/check/pipelines/.gitignore:
+ * tests/examples/volume/.gitignore:
+ ignore more
+ Original commit message from CVS:
+ ignore more
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * *.c, *.cc: don't mix tabs and spaces
+ configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
+ Original commit message from CVS:
+ 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ update core requirement to 0.10.4.1 because of async_playback
+ vmethod on GstBaseSink
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * 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
+ use DEBUG_FUNCPTR for collectpads
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
+ * gst/adder/gstadder.c: (gst_adder_init):
+ use DEBUG_FUNCPTR for collectpads
-2004-03-15 Johan Dahlin <johan@gnome.org>
+2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * *.h: Revert indenting
+ * Makefile.am:
+ don't go through check-torture if no check installed
+ Original commit message from CVS:
+ don't go through check-torture if no check installed
+
+2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
+ (gst_gnome_vfs_sink_class_init):
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
+ (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.c:
+ * ext/vorbis/vorbisenc.c:
+ * gst-libs/gst/audio/gstaudiofilter.c:
+ (gst_audio_filter_base_init):
+ * gst-libs/gst/audio/gstaudiofiltertemplate.c:
+ (gst_audio_filter_template_base_init):
+ * gst/adder/gstadder.c: (gst_adder_get_type):
+ * gst/adder/gstadder.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
+ (gst_audio_test_src_create):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_base_init):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
+ * gst/volume/gstvolume.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * tests/check/libs/cddabasesrc.c:
+ * tests/old/examples/gob/gst-identity2.gob:
+ Add docs for adder, use GST_ELEMENT_DETAILS macro,
+ define GstElementDetails at the top
-2004-03-15 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
+ Original commit message from CVS:
+ * win32/common/libgstinterfaces.def:
+ Add a lot of export functions for gst-python
+ * win32/common/libgstinterfaces.dsp:
+ Add a missing include folder in the project configuration
+
+2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create),
+ (gst_base_audio_src_change_state):
+ Fix audio sources, forgot to make the ringbuffer
+ startable...
+
+2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create),
+ (gst_base_audio_src_change_state):
+ unparent instead of unref the ringbuffer.
+
+2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
+ (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
+ Implement new async_play vmethod to start slaving and allow
+ playback start in case of async PLAY state changes.
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ Enable QoS with new method in base class.
+
+2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
+
+ gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
+ Original commit message from CVS:
+ Patch by: Julien MOUTTE <julien at moutte dot net>
+ * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
+ (gst_video_test_src_do_seek), (gst_video_test_src_create):
+ Partially handle 0 framerate, only EOS after the first frame
+ is missing.
+
+2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
+
+ gst/: Patch for support of YVU9 AVI files (#334822)
+ Original commit message from CVS:
+ Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
+ (gst_ffmpegcsp_avpicture_fill):
+ * gst/ffmpegcolorspace/imgconvert.c:
+ Patch for support of YVU9 AVI files (#334822)
+
+2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
+ Original commit message from CVS:
+ * docs/design/design-decodebin.txt:
+ Added design document for new decodebin
+ (Target Caps): text/x-pango-markup is also a default target caps.
+
+2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ docs/design/design-decodebin.txt: Added design document for new decodebin
+ Original commit message from CVS:
+ * docs/design/design-decodebin.txt:
+ Added design document for new decodebin
+
+2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_dispose):
+ Since we _parent the ringbuffer, we also need to
+ _unparent instead of a plain _unref.
+
+2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/seek.c: Add scrub checkbox.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
+ (stop_seek), (scrub_toggle_cb), (main):
+ Add scrub checkbox.
+
+2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
+ Original commit message from CVS:
+ * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
+ (gst_ogg_parse_chain):
+ Fix very inefficient usage of linked lists (#335365).
- * 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.
+2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gcc 4.1 unreferenced pointer fixes.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
+ * gst/playback/gstplaybin.c: (handoff):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property):
+ gcc 4.1 unreferenced pointer fixes.
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
+ gst_buffer_ref() now takes a GstBuffer*.
+
+2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
+ Original commit message from CVS:
+ 2006-03-20 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c:
+ (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
+ by Jan Schmidt.
+
+2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
+ (id3v1_type_find), (apetag_type_find), (plugin_init):
+ Can't do tag preferences via probability, as tags would then
+ lose against types that are recognised with MAXIMUM probability
+ (like .wav); so let all tag typefinders return MAXIMUM themselves
+ and order them via the rank. Split ID3v1 and ID3v2 typefinders so
+ that we can prefer APE to ID3v1 (fixes #335028).
-2004-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
- * *.c, *.h: run gst-indent
+ gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_change_state):
+ * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
+ (gst_ring_buffer_may_start):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Only start playback if we are playing.
+ should fix #330748.
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/modplug/gstmodplug.cc:
- * gst/modplug/gstmodplug.h:
- set correct timestamps on outgoing buffers
+ Revert accidental commits to these files.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
+ * win32/common/config.h:
+ Revert accidental commits to these files.
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
- * 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
+ tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
+ Original commit message from CVS:
+ Patch by: Michal Benes <michal dot benes at xeris dot cz>
+ * tests/Makefile.am:
+ Don't try to build tests in tests/icles if we
+ don't have X (#323852)
-2004-03-14 Benjamin Otte <otte@gnome.org>
+2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
- * 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
+ 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).
-2004-03-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * 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.
+ 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.
-2004-03-14 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/typefind/gsttypefindfunctions.c: (speex_type_find),
- (plugin_init):
- Add a typefind function for speex format
+2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-13 Ronald 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/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps),
- (gst_asf_demux_setup_pad):
- Use 25fps as our "fake" fps value (marked for fixage in 0.9.x)
- instead of 0. Reason is simple: some elements have a fps range
- of 1-max instead of 0-max. So now ASF video actually works.
+2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-03-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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.
- * po/LINGUAS:
- * po/sr.po:
- adding serbian as a language
+2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
-2004-03-13 Benjamin Otte <otte@gnome.org>
+ 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).
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- return taglist correctly from _get function, don't gst_pad_push it.
- (fixes #137042)
+2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
-2004-03-13 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/alsa/gstalsa.c: (gst_alsa_class_probe_devices):
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
+ Original commit message from CVS:
+ Patch by: Sebastien Moutte <sebastien moutte net>
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
+ (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
+ (gst_ffmpeg_smpfmt_to_caps):
+ Replace __VA_ARGS__ caps creation macros with varargs functions.
+ Makes things compile on MSVC (#320765), looks nicer, and we can
+ tell the compiler to check for the NULL terminator.
-2004-03-13 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
- * 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
+ gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
+ Original commit message from CVS:
+ Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make sure the buffer we copy into is really always big
+ enough, this time for real (#333488).
- * configure.ac:
- use pkg-config for some libraries, falling back to the old .m4 way
- (fixes #131270)
- * m4/libdv.m4:
- removed
+2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-10 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Add support for 24bpp DIB (#305279).
- * configure.ac:
- * tools/Makefile.am:
- * tools/Makefile.in:
- * tools/gst-launch-ext-m.m:
- * tools/gst-launch-ext.1.in:
- * tools/gst-visualise-m.m:
- * tools/gst-visualise.1:
- * tools/gst-visualise.1.in:
- reorganizing generation of script tools
+2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-03-10 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Re-enable QoS after the release.
+ Original commit message from CVS:
+ * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
+ * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
+ (gst_video_scale_init), (gst_video_scale_src_event):
+ Re-enable QoS after the release.
+ Rework videoscale to use the base class src_event handler.
- * ext/divx/gstdivxdec.c:
- Downgrade priority. We prefer ffdec_mpeg4.
- * ext/faad/gstfaad.c: (gst_faad_srcgetcaps), (gst_faad_srcconnect),
- (gst_faad_chain), (gst_faad_change_state):
- Fix capsnego. Doesn't work for some sounds because we don't have
- a 5:1 to stereo element.
- * ext/xvid/gstxvid.c: (plugin_init):
- Add priority.
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- Add discont handling.
+2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Colin Walters <walters@verbum.org>
+ configure.ac: back to CVS.
+ Original commit message from CVS:
+ * configure.ac:
+ back to CVS.
- * gst/audioconvert/gstaudioconvert.c: Fix typo in width 8
- conversion.
+=== release 0.10.5 ===
-2004-03-09 Benjamin Otte <otte@gnome.org>
+2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init):
- the signals take 2 arguments
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ releasing 0.10.5
+ Original commit message from CVS:
+ releasing 0.10.5
-2004-03-09 David Schleef <ds@schleef.org>
+2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad),
- (gst_alsa_fixate): Add fixate function. (bug #136686)
- * ext/alsa/gstalsa.h:
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_init):
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
-2004-03-09 Benjamin Otte <otte@gnome.org>
+2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init), (gst_mikmod_loop),
- (gst_mikmod_change_state):
- * ext/mikmod/gstmikmod.h:
- make mikmod's loop function not loop infinitely and call
- gst_element_yield anymore
- * gst/modplug/gstmodplug.cc:
- fix pad negotiation (fixes #136590)
+ docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ Part of previous cdparanoiasrc docs fixes, forgot to commit.
-2004-03-09 David Schleef <ds@schleef.org>
+2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/lcs/Makefile.am: Fix so that the lcs colorspace plugin
- doesn't conflict with the internal colorspace plugin.
- * gst-libs/gst/audio/make_filter: Use `` instead of $() to
- satisfy the crappy-ass shell shipped by a certain vendor.
- * gst/videofilter/make_filter: same (bug #135299)
+ docs/plugins/: Add cdparanoiasrc to docs.
+ Original commit message from CVS:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ Add cdparanoiasrc to docs.
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ More GstCddaBaseSrc docs.
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: bump nano to 1
+ Add new API to libgsttag: gst_tag_from_id3_user_tag().
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
+ * gst-libs/gst/tag/tag.h:
+ Add new API to libgsttag: gst_tag_from_id3_user_tag().
-=== release 0.7.6 ===
+2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (plugin_init):
+ NULL-terminate array of mpeg4 video file extensions.
+ Fixes crash on PPC (#334226).
- * configure.in: releasing 0.7.6, "There"
+2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ (gst_gnome_vfs_src_check_get_range):
+ gnome_vfs_uri_is_local() alone is not a good indicator
+ whether we can operate in pull-mode with a specific URI,
+ as it returns FALSE for file:// URIs that point to an
+ NFS-mounted path. Be more conservative here: whitelist
+ local files, blacklist http URIs and use the old
+ mechanism for anything else (fixes #334216).
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- * pkgconfig/gstreamer-play.pc.in:
- synchronize the two
+2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+ configure.ac: back to trunk
+ Original commit message from CVS:
+ * configure.ac:
+ back to trunk
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_base_init),
- (cdparanoia_open), (cdparanoia_event):
- fix/add error handling
- * po/POTFILES.in:
- add cdparanoia source
- * tools/Makefile.am:
- make scripts executable
+=== release 0.10.4 ===
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- * ext/vorbis/Makefile.am:
- * sys/Makefile.am:
- remove id3types, vorbisfile and xvideosink from the build (#133783)
+ * 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
-2004-03-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- Fix metadata read crash (#136537).
+ 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.
-2004-03-08 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * 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
+ gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Make sure we don't read beyond the palette buffer in case of
+ broken or manipulated files (#333488, patch by: Fabrizio
+ Gennari)
-2004-03-08 Christian Schaller <Uraeus@gnome.org>
+2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
- * 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>
+ 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.
- * sys/oss/gstosssink.c: (gst_osssink_get_delay),
- (gst_osssink_get_time):
- add a warning, IMO this won't get triggered anymore, remove later
+2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-07 David Schleef <ds@schleef.org>
+ * configure.ac:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * win32/common/config.h:
+ prereleasing
+ Original commit message from CVS:
+ prereleasing
+
+2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/libvisual/visual.c: Small cleanups.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (gst_visual_get_type),
+ (gst_visual_src_setcaps), (gst_vis_src_negotiate),
+ (gst_visual_chain):
+ Small cleanups.
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c: (gst_theora_dec_init),
+ (gst_theora_dec_reset), (_theora_granule_time),
+ (theora_dec_src_convert), (theora_dec_sink_convert),
+ (theora_dec_src_query), (theora_dec_src_event),
+ (theora_dec_sink_event), (theora_handle_comment_packet),
+ (theora_handle_header_packet), (theora_dec_push),
+ (theora_handle_data_packet), (theora_dec_chain),
+ (theora_dec_change_state):
+ Add simple QoS.
+
+2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
+ (audiocast_register_listener), (gst_gnome_vfs_src_start):
+ Some cleanups.
+
+2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
+ Don't try to activate NULL chains.
+
+2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
+ Fix invalid memory access to region before peek'd data (#332964).
- * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Added Cinepak
- format (bug #136470)
+2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
-2004-03-07 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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-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
+2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-07 Christian Schaller <Uraeus@gnome.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.
- * ext/nas/nassink.c:
- A bunch of NAS fixes from Arwed von Merkatz
+2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-06 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).
- * 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.
+2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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).
- * 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.
+2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
+ 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.
- * 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...
+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.
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * gst-libs/gst/media-info/media-info-priv.c: (have_type_callback),
- (deep_notify_callback), (tag_flag_score), (found_tag_callback),
- (error_callback), (gmi_reset), (gmi_seek_to_track),
- (gmi_get_decoder), (gmi_set_mime), (gmip_find_type_pre),
- (gmip_find_type_post), (gmip_find_stream_post),
- (gmip_find_track_streaminfo_post):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c: (print_tag),
- (info_print), (main):
- * gst-libs/gst/media-info/media-info.c:
- (gst_media_info_error_create), (gst_media_info_error_element),
- (gst_media_info_instance_init), (gst_media_info_get_property),
- (gst_media_info_new), (gst_media_info_set_source),
- (gst_media_info_read_idler), (gst_media_info_read):
- * gst-libs/gst/media-info/media-info.h:
- first pass at making this work again. This seems to work on
- tagged ogg/vorbis and mp3 files.
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- fix huge leak: gst_buffer_merge doesn't unref the first argument
- itself.
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/mad/gstmad.c: (gst_mad_class_init), (gst_mad_update_info):
- report layer/mode/emphasis
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * ext/mad/gstmad.c: (gst_mad_chain): fixed caps leak
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_new):
- signal serial
-
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/vorbis/vorbis.c: (plugin_init):
- * ext/vorbis/vorbisdec.c: (vorbis_dec_get_formats),
- (gst_vorbis_dec_init), (vorbis_dec_event):
- add debug category
- make vorbisdec handle _BYTE and _TIME queries
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- * ext/mad/gstmad.c: (gst_mad_chain): send the average bitrate read
- from the xing header
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
- (gst_audio_convert_link), (gst_audio_convert_change_state),
- (gst_audio_convert_buffer_from_default_format):
- do conversions from/to float correctly, fix some caps nego errors,
- export correct supported caps in template and getcaps, use correct
- caps in try_set_caps functions
-
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
-
- For some reason, I only committed a ChangeLog entry yesterday and
- not the corresponding code...
- * ext/mad/gstmad.c: Fix detection of Xing headers
- * gst/tags/gstid3tag.c: Changes to support TLEN tags
-
-2004-03-06 Benjamin Otte <otte@gnome.org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_get_pad_by_pad),
- (gst_ogg_demux_src_query):
- make sure to handle the case where there's no current chain
- gracefully.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * ext/aalib/gstaasink.c: (gst_aasink_fixate), (gst_aasink_init):
- Add fixate function. (bug #131128)
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
- (gst_sdlvideosink_fixate): Add fixate function.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Fix attempt to print a non-pointer using GST_PTR_FORMAT.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt):
- Fix missing break that was causing ulaw to be interpreted as
- raw int.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice):
- Fix code that ignores return value of gst_buffer_merge().
- (bug #114560)
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_segment):
- * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_add_slice): same
- * testsuite/gst-lint: Check for above.
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * gst/udp/gstudpsrc.c: (gst_udpsrc_get): Check for unfixed
- caps and throw an element error. (bug #136334)
-
-2004-03-05 David Schleef <ds@schleef.org>
-
- * ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
- (gst_faad_chain): Fix negotiation.
- * ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
- key and button events.
- * gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
- dung heap of code.
- * gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
- depends on gconf
- * gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
- function to encourage better negotiation, particularly between
- audioconvert and osssink.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
- * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
- more important.
- * gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
- typefinding.
- * gst/vbidec/vbiscreen.c: Add glib header
- * pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
+ gst-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).
-2004-03-06 Christophe Fergeau <teuf@users.sourceforge.net>
+2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mad/gstmad.c: Fix detection of Xing headers
- * gst/tags/gstid3tag.c: Changes to support TLEN tags
+ 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().
-2004-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_pad_convert), (gst_wavparse_pad_query):
- debug updates
+ 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().
-2004-03-06 Christophe Fergeau <teuf@gnome.org>
+2006-03-07 12:49:03 +0000 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
+ 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).
-2004-03-05 Benjamin Otte <otte@gnome.org>
+2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
- convert channels correctly. convert correctly to unsigned.
+ 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
-2004-03-05 Julien MOUTTE <julien@moutte.net>
+2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state): Check if
- we have a window before clearing it.
+ 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).
-2004-03-05 Julien MOUTTE <julien@moutte.net>
+2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
- * sys/ximage/ximagesink.c: (gst_ximagesink_change_state): Check if we
- have a window before clearing it.
+ 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.
-2004-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
- * 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
+ 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).
-2004-03-05 David Schleef <ds@schleef.org>
+2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.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.
+ 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.
-2004-03-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * 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.
+ * ext/ogg/gstoggmux.c:
+ changed more than 5 lines
+ Original commit message from CVS:
+ changed more than 5 lines
-2004-03-05 Benjamin Otte <otte@gnome.org>
+2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.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>
+ 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
- * configure.ac:
- fix --with-plugins, don't think it ever worked before
- * gst-plugins.spec.in:
- even more updates
+2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
-2004-03-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ 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)
- * ext/sdl/sdlvideosink.h:
- * sys/ximage/ximagesink.h:
- * sys/xvideo/xvideosink.h:
- * sys/xvimage/xvimagesink.h:
- Fix for move of gstvideosink.h -> videosink.h.
+2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-02 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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).
- * gst-libs/gst/xwindowlistener/Makefile.am:
- this is a plugin library, not a library
+2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-01 David Schleef <ds@schleef.org>
+ 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).
- * AUTHORS: Added some names. Add yourself if you're still
- missing.
+2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-03-01 David Schleef <ds@schleef.org>
+ 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).
- * MAINTAINERS: Add
+2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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-plugins.spec.in: clean up spec file
+2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/ogg/README:
+ * ext/ogg/gstoggmux.c:
+ debug updates
+ Original commit message from CVS:
+ debug updates
- * 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
+2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ 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/tags/Makefile.am: fix plugin filename
+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>
-2004-03-01 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst/videorate/Makefile.am:
+ fix wim's commit
+ Original commit message from CVS:
+ fix wim's commit
- * examples/gstplay/player.c: (got_time_tick), (main):
- add error handler
- display time_tick more readably
- * gst/mixmatrix/Makefile.am:
- fix plugin file name
+2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-29 Christophe Fergeau <teuf@gnome.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/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
+ * ext/ogg/gstoggmux.c:
+ clean up debug messages
+ Original commit message from CVS:
+ clean up debug messages
-2004-02-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_init),
- (gst_videodrop_chain), (gst_videodrop_change_state):
- * gst/videodrop/gstvideodrop.h:
- Work based on timestamp of input data, not based on the expected
- framerate from the input. The consequence is that this element now
- not only scales framerates, but also functions as a framerate
- corrector or framerate stabilizer/constantizer.
+ * ext/ogg/gstoggmux.c:
+ extra debugging from older version, makes it easier to compare
+ Original commit message from CVS:
+ extra debugging from older version, makes it easier to compare
-2004-02-27 David Schleef <ds@schleef.org>
+2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- patches from jmmv@menta.net (Julio M. Merino Vidal)
+ * 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.
- * 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.
+2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-27 Benjamin Otte <otte@gnome.org>
+ ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
+ (gst_ogm_parse_chain):
+ Don't remove/re-add source pad if the new caps are the same as
+ the old caps anyway (#333042). When removing source pad, don't
+ unref it afterwards - we didn't ref it when adding. Sprinkle some
+ GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
+ after using gst_pad_get_parent(). Return downstream flow return
+ value in chain function.
- * gst-libs/gst/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.
+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)
-2004-02-26 David Schleef <ds@schleef.org>
+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.
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- * gst/level/Makefile.am: -marshal.[ch] and -enum.[ch] files
- should not be disted, -marshal.h files should not be installed,
- and -enum.h files _should_ be installed. Fix to make this the
- case.
+2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
-=== release 0.7.5 ===
+ ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
+ (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
+ (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
+ (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
+ (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
+ (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
+ Annodex support in ogg demuxer. Doesn't do very much without the
+ other annodex patches (to come).
-2004-02-26 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: release 0.7.5, "Under The Sea"
+ gst-libs/gst/riff/riff-media.c:
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
+ Pick up palette for MS video v1 (#327028, patch by:
+ Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
- (gst_audio_convert_change_state), (gst_audio_convert_get_buffer):
- * gst/videoscale/gstvideoscale.c:
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- assorted debug/warning fixes
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ (gst_ffmpegcsp_caps_remove_format_info),
+ (gst_ffmpegcsp_get_unit_size):
+ The 'palette_data' field from incoming RGB caps shouldn't be
+ proxied on outgoing YUV caps; also, restrict unit size
+ adjustment in case of paletted data only to the unit that
+ actually has a palette. Fixes #330711.
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps),
- (gst_videoscale_init), (gst_videoscale_chain),
- (gst_videoscale_set_property), (plugin_init):
- * gst/videoscale/gstvideoscale.h:
- * gst/videoscale/videoscale.c: (gst_videoscale_setup),
- (gst_videoscale_scale_rgb), (gst_videoscale_planar411),
- (gst_videoscale_planar400), (gst_videoscale_packed422),
- (gst_videoscale_packed422rev), (gst_videoscale_32bit),
- (gst_videoscale_24bit), (gst_videoscale_16bit),
- (gst_videoscale_bilinear), (gst_videoscale_bicubic),
- (gst_videoscale_scale_plane_slow),
- (gst_videoscale_scale_point_sample),
- (gst_videoscale_scale_nearest),
- (gst_videoscale_scale_nearest_str2),
- (gst_videoscale_scale_nearest_str4),
- (gst_videoscale_scale_nearest_32bit),
- (gst_videoscale_scale_nearest_24bit),
- (gst_videoscale_scale_nearest_16bit):
- add debugging category and use it properly
- fix use of GST_PTR_FORMAT
+ gst/ffmpegcolorspace/gstffmpegcolorspace.c: 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.
-2004-02-25 Andy Wingo <wingo@pobox.com>
+2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/interleave/interleave.c (interleave_buffered_loop): Always
- push only when channel->buffer is NULL. Prevents segfaults doing
- the state change after a nonlocal exit, like a scheme exception.
+ 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/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
- Handle the case where the intersected caps is empty.
+2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-25 Thomas Vander Stichele <thomas at apestaart dot org>
+ * 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/law/mulaw-decode.c: (mulawdec_link):
- * gst/law/mulaw.c: (plugin_init):
- fix mulawdec so it actually works again
+2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
-2004-02-24 Arwed v. Merkatz <v.merkatz@gmx.net>
+ 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.
- reviewed by: David Schleef <ds@schleef.org>
+2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/videofilter/gstgamma.c: (gst_gamma_class_init),
- (gst_gamma_init), (gst_gamma_set_property),
- (gst_gamma_get_property), (gst_gamma_calculate_tables),
- (gst_gamma_rgb24), (gst_gamma_rgb32): Adds gamma correction
- for RGB, with separate r g and b correction factors. (#131167)
+ gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
+ Original commit message from CVS:
+ Reviewed by : Edward Hervey <edward@fluendo.com>
+ * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
+ (qt_type_find):
+ Better 3gp typefinding.
-2004-02-24 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
- only signal tags for bitrate if they're > 0 (#134894)
+ ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
+ Don't send EOS event here, the base class will send one for us.
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Subpictures without video stream aren't allowed either.
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
+ Fix debug statement copy'n'paste-o.
-2004-02-24 David Schleef <ds@schleef.org>
+2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/qtdemux/qtdemux.c: (plugin_init), (gst_qtdemux_loop_header),
- (qtdemux_parse_moov), (qtdemux_parse), (qtdemux_node_dump_foreach),
- (qtdemux_dump_mvhd), (qtdemux_dump_tkhd), (qtdemux_dump_elst),
- (qtdemux_dump_mdhd), (qtdemux_dump_hdlr), (qtdemux_dump_vmhd),
- (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_stts),
- (qtdemux_dump_stss), (qtdemux_dump_stsc), (qtdemux_dump_stsz),
- (qtdemux_dump_stco), (qtdemux_dump_co64), (qtdemux_dump_dcom),
- (qtdemux_dump_cmvd), (qtdemux_parse_tree), (qtdemux_parse_trak):
- Cleanups. Convert g_prints to GST_LOGs. Add qtdemux debug
- category. Attempt to fix timestamp calculation.
+ ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
+ Original commit message from CVS:
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
+ Fix issues with mixer keeping state when muting/unmuting
+ and when changing the volume whilst muted (see #331763
+ and #331765).
-2004-02-24 Johan Dahlin <johan@gnome.org>
+2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/gconf/gconf.c: Add \n to g_print error messages
+ 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).
-2004-02-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * gconf/Makefile.am:
- * gconf/gstreamer.schemas:
- * gst-libs/gst/gconf/Makefile.am:
- * gst-libs/gst/gconf/gconf.c:
- version gconf schemas and install locations
-
-2004-02-23 Benjamin Otte <otte@gnome.org>
-
- * ext/xine/xineinput.c: (gst_xine_input_dispose):
- (gst_xine_input_subclass_init):
- call parent dispose.
- change pad template for CD reader correctly
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c: (plugin_init):
- * ext/xine/xineaudiosink.c:
- wrap audio sinks, too
- * gst-libs/gst/resample/private.h:
- * gst-libs/gst/resample/resample.c: (gst_resample_init),
- (gst_resample_reinit), (gst_resample_scale),
- (gst_resample_nearest_s16), (gst_resample_bilinear_s16),
- (gst_resample_sinc_slow_s16), (gst_resample_sinc_s16),
- (gst_resample_sinc_ft_s16), (gst_resample_nearest_float),
- (gst_resample_bilinear_float), (gst_resample_sinc_slow_float),
- (gst_resample_sinc_float), (gst_resample_sinc_ft_float):
- * gst-libs/gst/resample/resample.h:
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
- (gst_audioscale_class_init), (gst_audioscale_link),
- (gst_audioscale_get_buffer), (gst_audioscale_init),
- (gst_audioscale_chain), (gst_audioscale_set_property),
- (gst_audioscale_get_property):
- * gst/audioscale/gstaudioscale.h:
- s/resample_*/gst_resample_*/i to not clobber namespaces
-
-2004-02-23 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
- (gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
- (gst_riff_create_video_template_caps),
- (gst_riff_create_audio_template_caps),
- (gst_riff_create_iavs_template_caps):
- * gst-libs/gst/riff/riff-media.h:
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
- (gst_asf_demux_audio_caps), (gst_asf_demux_add_audio_stream),
- (gst_asf_demux_video_caps), (gst_asf_demux_add_video_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
- (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
- (gst_matroska_demux_plugin_init): First batch implementing audio and
- video codec tags in demuxers.
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c: (plugin_init):
- * ext/xine/xineinput.c:
- add input plugin wrapper. Playback from files, http, mms and cdda
- works.
- * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
- remove leftover G_GNUC_UNUSED
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_stream),
- (gst_asf_demux_identify_guid):
- improve debugging output
-
-2004-02-22 Benjamin Otte <otte@gnome.org>
-
- reported by: Padraig O'Briain <padraig.obriain@sun.com>
+ gst-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.
- * autogen.sh:
- replace test -e with test -x for mkinstalldirs to be more portable.
- (fixes #134816)
+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).
-2004-02-22 Benjamin Otte <otte@gnome.org>
+2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
- reported by: Stefan Kost <kost@imn.htwk-leipzig.de>
+ 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.
- * gst/audioconvert/gstaudioconvert.c: (plugin_init):
- set rank to PRIMARY
- * gst/volume/gstvolume.c: (plugin_init):
- set rank to NONE
- fixes #134960
+2006-02-19 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>
-2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
+ * 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).
- reviewed by Benjamin Otte <otte@gnome.org>
+2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
+ (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
+ (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
+ (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
+ (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
+ (gst_ring_buffer_pause), (gst_ring_buffer_stop),
+ (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
+ (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
+ (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
+ (gst_ring_buffer_clear):
+ Small cleanups.
+ Added some G_LIKELY.
+
+2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/TODO: Update TODO
+ Original commit message from CVS:
+ * gst-libs/gst/audio/TODO:
+ Update TODO
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_offset):
+ When trying to play samples ASAP and we don't have a
+ previous sample, try to play at position 0 instead of
+ an invalid position.
+
+2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_open),
+ (gst_alsasink_reset):
+ Also release lock when we get an error in _reset();
+ fix an error message.
+
+2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
+ (gst_alsasink_init), (get_channel_free_structure),
+ (caps_add_channel_configuration), (gst_alsasink_getcaps),
+ (gst_alsasink_close):
+ * ext/alsa/gstalsasink.h:
+ Add support for more than 2 channels (#326720).
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- escape NULL strings in GST_ELEMENT_ERROR properly (fixes #135116)
+2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-22 Benjamin Otte <otte@gnome.org>
+ gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
+ Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
+ with 4 or 6 channels, assume a default channel layout to make things
+ work (not sure there's anything else we can do in those cases).
- * configure.ac:
- export [_]*{gst,Gst,GST}.* symbols from plugins
+2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-22 Christophe Fergeau <teuf@gnome.org>
+ 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.
- reviewed by: Benjamin Otte <otte@gnome.org>
+2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/lame/gstlame.c: (add_one_tag):
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value),
- (gst_vorbisenc_metadata_set1):
- * gst/tags/gstid3tag.c:
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add):
- apply fixes from bugs #135042 (lame can't write tags) and #133817
- (add GST_ALBUM_VOLUME_{COUNT,NUMBER} tags)
+ gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
+ Original commit message from CVS:
+ Reviewed by Edward Hervey <edward@fluendo.com>
+ * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
+ C-level optimization of the RGBA nearest neighbour function.
+ Eventually this might end up in liboil with vectorized versions.
-2004-02-22 Ramon Garcia <ramon_garcia_f@yahoo.com>
+2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac: Export only gst_plugin_desc from plugins.
- Note that this change only makes any effect with Linux using libtool
- 1.5.2 or higher. Otherwise it is silently ignored, but it would build
- fine. And don't try to have several versions of libtool in different
- directories.
+ gst-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).
-2004-02-20 Andy Wingo <wingo@pobox.com>
+2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
- interleave respectively.
+ 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.
- * gst/interleave/deinterleave.c: New plugin: deinterleave
- (replaces on oneton).
- * gst/interleave/interleave.c: New plugin: interleave.
- * gst/interleave/plugin.h: Support file.
- * gst/interleave/plugin.c: Support file.
+2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
- * configure.ac: Remove intfloat and oneton, add interleave.
+ 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.
- * ext/sndfile/gstsf.c: Handle events better.
+2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
- * gst/audioconvert/gstaudioconvert.c: Change to support int2float
- and float2int operation. int2float has scheduling problems as
- noted in in2float_chain.
+ gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
+ quicktime movie files can also contain 'uuid' atoms.
-2004-02-20 Benjamin Otte <otte@gnome.org>
+2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/xine/Makefile.am:
- * ext/xine/gstxine.h:
- * ext/xine/xine.c:
- * ext/xine/xineaudiodec.c:
- * ext/xine/xinecaps.c:
- add first version of xine plugin wrapper. Currently only wraps the
- QDM2 win32 DLL, and even that only in proof-of-concept quality.
- * configure.ac:
- * ext/Makefile.am:
- add xine plugin wrapper, disabled by default. Use --enable-xine to
- build. Note that it'll segfault on gst-register if you don't remove
- the goom and tvtime post plugins from xine.
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
- (qtdemux_parse), (qtdemux_parse_trak), (qtdemux_audio_caps):
- add extradata parsing for QDM2.
- change around debugging prints.
+ gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
+ Original commit message from CVS:
+ * gst/audioconvert/plugin.c: (plugin_init):
+ Register the GstAudioChannelPosition enum type with the type
+ system in the plugin_init function, so that it is known before
+ any element actually makes use of multi-channel stuff. This is
+ required for example if one wants to be able to deserialise/use
+ a caps string with channel positions before any pipeline has
+ been setup and started, like with gst-launch.
-2004-02-19 Benjamin Otte <otte@gnome.org>
+2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * ext/lame/gstlame.c: (gst_lame_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- use gst_tag_list_insert when you want to insert tags
+ gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
+ (gst_ring_buffer_samples_done), (wait_segment),
+ (gst_ring_buffer_commit), (gst_ring_buffer_clear):
+ Add some compiler G_(UN_)LIKELY help.
+ SIGNAL the ringbuffer waiters when going to PAUSED as well to
+ make sure they can exit their functions. Should fix #330748
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * configure.ac: Move massink to gst-rotten
- * ext/Makefile.am:
- * ext/mas/Makefile.am:
- * ext/mas/massink.c:
- * ext/mas/massink.h:
+ Windows does not have long long; copy the generated _stdint.h
+ Original commit message from CVS:
+ * Makefile.am:
+ * configure.ac:
+ * win32/MANIFEST:
+ * win32/common/_stdint.h:
+ Windows does not have long long; copy the generated _stdint.h
+ * win32/common/interfaces-enumtypes.c:
+ (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
+ (gst_mixer_track_flags_get_type),
+ (gst_tuner_channel_flags_get_type):
+ * win32/common/multichannel-enumtypes.c:
+ (gst_audio_channel_position_get_type):
+ update
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
- * 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.
+ gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Always sync on first sample we receive when starting.
-2004-02-18 David Schleef <ds@schleef.org>
+2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
- * gst/sine/demo-dparams.c: (quit_live),
- (dynparm_log_value_changed), (dynparm_value_changed), (main):
- Use double dparams, not float.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_init): Change sync default to FALSE, since multiple
- sync'd elements don't really work correctly.
- * gst/volume/gstvolume.c: (volume_class_init), (volume_init),
- (volume_update_volume), (volume_get_property): Change dparam
- to double.
+ gst/playback/gstplaybin.c: Update vis bin docs.
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_vis_element):
+ Update vis bin docs.
+ Move queue after tee so we don't queue video buffers but
+ audio samples instead. Fixes problems where the video queue
+ is filled and the audio queue empty.
-2004-02-18 Julien MOUTTE <julien@moutte.net>
+2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.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.
+ gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
+ No need to push an EOS event here, GstBaseSrc will do that for us
+ when we return FLOW_UNEXPECTED.
-2004-02-15 Julien MOUTTE <julien@moutte.net>
+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
- * 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.
+2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-14 Julien MOUTTE <julien@moutte.net>
+ * ChangeLog:
+ changelog foo
+ Original commit message from CVS:
+ changelog foo
- * 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.
+2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * 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
- * 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>
+=== release 0.10.3 ===
- * 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>
+2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- * ext/Makefile.am:
- * gst-libs/ext/Makefile.am:
- move ffmpeg plugin to gst-ffmpeg module
-
-2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * win32/common/config.h:
+ Releasing 0.10.3
+ Original commit message from CVS:
+ Releasing 0.10.3
- * configure.ac: use GST_ARCH to detect architecture
+2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-12 Julien MOUTTE <julien@moutte.net>
+ configure.ac: Drat. Bump libtool version number for new API.
+ Original commit message from CVS:
+ * configure.ac:
+ Drat. Bump libtool version number for new API.
+ Prelease 0.10.2.3 (of 0.10.3)
- * gst/vbidec/vbiscreen.c: Fixing thomasvs fixes. Missing header.
+2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
+ 0.10.2.2 prerelease (of 0.10.3).
+ Original commit message from CVS:
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.2.2 prerelease (of 0.10.3).
- * ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
- classify LADSPA plugins based on number of src/sink pads
- (#133663, Stefan Kost)
- * gst/sine/gstsinesrc.c: (gst_sinesrc_init):
- fix dparams registration
- (#133528, Stefan Kost)
- * gst/vbidec/vbiscreen.c: (vbiscreen_set_current_cell):
- fix use of isprint and use g_ascii_isprint instead
- (#133316, Stefan Kost)
+2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
+ Original commit message from CVS:
+ * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
+ Revert Andy's newsegment change pending a more correct
+ fix.
- Convert a few inner loops to use liboil. This is currently
- optional, and is only enabled if liboil is present (duh!).
- * configure.ac: Check for liboil-0.1
- * gst/intfloat/Makefile.am:
- * gst/intfloat/gstint2float.c: (conv_f32_s16), (scalarmult_f32),
- (gst_int2float_chain_gint16):
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_class_init),
- (tablelookup_u8), (gst_videobalance_planar411):
- * gst/videotestsrc/Makefile.am:
- * gst/videotestsrc/gstvideotestsrc.c: (plugin_init):
- * gst/videotestsrc/videotestsrc.c: (splat_u8), (paint_hline_YUY2),
- (paint_hline_IYU2), (paint_hline_str4), (paint_hline_str3),
- (paint_hline_RGB565), (paint_hline_xRGB1555):
+2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ Update .po files
+ Original commit message from CVS:
+ Update .po files
- * ext/lcs/gstcolorspace.c: (colorspace_find_lcs_format),
- (gst_colorspace_caps_get_fourcc), (colorspace_setup_converter),
- (gst_colorspace_getcaps), (gst_colorspace_link),
- (gst_colorspace_base_init), (gst_colorspace_init),
- (gst_colorspace_chain), (gst_colorspace_change_state),
- (plugin_init): Merge Ronald's patch (bug #117897) and update
- for new caps and negotiation. Seems to work, although it
- shows off bugs in lcs.
+2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-11 David Schleef <ds@schleef.org>
+ * gst/tcp/gstmultifdsink.c:
+ doc fixes
+ Original commit message from CVS:
+ doc fixes
- * ext/alsa/Makefile.am: Fix linking against libgstinterfaces.
- (bug #133886) Noticed by bugs@leroutier.net (Stephane LOEUILLET)
+2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
+ Original commit message from CVS:
+ :
+ * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
+ (qt_type_find), (plugin_init):
+ detect more files as 3gp
+ group and reorder the iso file formats
- * ext/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init),
- (gst_rfbsrc_change_state), (gst_rfbsrc_init),
- (gst_rfbsrc_set_property), (gst_rfbsrc_get_property):
- Add server and port properties
+2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
+ Original commit message from CVS:
+ * ext/vorbis/vorbis.c: (plugin_init):
+ Register musicbrainz tags, so apps don't have to.
- * m4/a52.m4:
- * m4/aalib.m4:
- * m4/as-ffmpeg.m4:
- * m4/as-liblame.m4:
- * m4/as-slurp-ffmpeg.m4:
- * m4/check-libheader.m4:
- * m4/esd.m4:
- * m4/freetype2.m4:
- * m4/gconf-2.m4:
- * m4/glib.m4:
- * m4/gst-alsa.m4:
- * m4/gst-artsc.m4:
- * m4/gst-ivorbis.m4:
- * m4/gst-matroska.m4:
- * m4/gst-sdl.m4:
- * m4/gst-shout2.m4:
- * m4/gst-sid.m4:
- * m4/gtk.m4:
- * m4/libdv.m4:
- * m4/libfame.m4:
- * m4/libmikmod.m4:
- * m4/ogg.m4:
- * m4/vorbis.m4:
- fix underquotedness of macros (#133800)
- * m4/as-avifile.m4:
- * m4/xmms.m4:
- removed because no longer used
+2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
+ (gst_tag_to_vorbis_tag):
+ Make sure we called gst_tag_register_musicbrainz_tags()
+ before possibly mapping a vorbiscomment string from/to a
+ musicbrainz tag.
- * configure.ac:
- require gettext 0.11.5 so ulonglong.m4 gets checked out and copied
- by autopoint (fixes #132996)
+2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
+ In case we can't find the required number of consecutive
+ mpeg audio frames to positively identify an MPEG audio
+ stream, check if there's at least a valid mpeg audio
+ frame right at offset 0 and if so suggest mpeg/audio
+ caps with a very low probability (#153004).
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_base_init):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_base_init):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_base_init):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_base_init):
- fix memleaks
+2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-11 David Schleef <ds@schleef.org>
+ gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
+ Original commit message from CVS:
+ 2006-02-07 Andy Wingo <wingo@pobox.com>
+ * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
+ a TIME segment if we get timestamped buffers. Requires recent
+ fixes in core to work properly.
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_chain): Fix logic bug causing spurious errors.
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_base_init),
- (gst_jpegdec_init), (gst_jpegdec_chain): Fix negotiation.
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_base_init),
- (gst_jpegenc_class_init), (gst_jpegenc_init),
- (gst_jpegenc_getcaps), (gst_jpegenc_link), (gst_jpegenc_resync),
- (gst_jpegenc_chain), (gst_jpegenc_set_property),
- (gst_jpegenc_get_property): Fix negotiation. Add some properties.
- * ext/jpeg/gstjpegenc.h: Fix negotiation.
+2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-10 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Don't print the URI as part of the error message, it
+ makes error dialogs look rather ugly, especially if
+ the URI is very long or has characters in it that
+ need escaping.
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init),
- (gst_mikmod_srcfixate), (gst_mikmod_srclink), (gst_mikmod_loop):
- * ext/mikmod/gstmikmod.h:
- fix caps negotiation in mikmod
- * ext/ogg/gstoggdemux.c: (gst_ogg_print):
- output debug information
+2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-08 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (prepare_output):
+ Error out if we have only text or subtitles, but nothing
+ else. Also error out if we have subtitles but no video
+ stream.
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/navigation/Makefile.am:
- * gst-libs/gst/xoverlay/Makefile.am:
- remove unused GST_OPT_CFLAGS from Makefiles
- include X_CFLAGS and X_LIBS in xoverlay. (#131948)
+2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-07 David Schleef <ds@schleef.org>
+ ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
+ Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
+ Post an error message on the bus when we encounter an
+ error, which will hopefully be more meaningful than the
+ 'Internal Flow Error' message users get to see if we
+ just return GST_FLOW_ERROR.
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_handle_event): Don't
- push events to pads that haven't been created (#133508)
+2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
-2004-02-07 Jan Schmidt <thaytan@mad.scientist.com>
+ configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
+ Original commit message from CVS:
+ 2006-02-07 Andy Wingo <wingo@pobox.com>
+ * configure.ac (GST_MAJORMINOR): Update core version req to
+ 0.10.2.2, for the collectpads API addition (#330244).
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_src_convert),
- (gst_dvdec_sink_convert), (gst_dvdec_handle_sink_event),
- (gst_dvdec_video_getcaps), (gst_dvdec_video_link),
- (gst_dvdec_loop), (gst_dvdec_change_state):
- Second attempt at committing a working dvdec element.
+2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-06 David Schleef <ds@schleef.org>
+ ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ Return FALSE from plugin_init() when GnomeVFS can't
+ be initialised for some reason (#328423).
- Build fixes for OS X: (see #129600)
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strh),
- (gst_riff_read_strf_vids), (gst_riff_read_strf_auds),
- (gst_riff_read_strf_iavs):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml):
- * gst/playondemand/Makefile.am:
- * gst/rtp/rtp-packet.c:
+2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
-2004-02-05 David Schleef <ds@schleef.org>
+ ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
+ Original commit message from CVS:
+ 2006-02-06 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
+ Stick to seeking theory until i find the bug.
+ * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_loop): Revert
- last change, because it Just Doesn't Compile.
+2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ Make theoraenc and the tests leak free. Like, really.
+ Original commit message from CVS:
+ * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
+ (theora_enc_finalize), (theora_enc_sink_setcaps),
+ (theora_set_header_on_caps), (theora_enc_chain),
+ (theora_enc_change_state):
+ * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
+ Make theoraenc and the tests leak free. Like, really.
- * ext/mad/gstid3tag.c: (gst_id3_tag_chain):
- skip undecodable id3v2 tag instead of keeping it
+2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 David Schleef <ds@schleef.org>
+ Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
+ Original commit message from CVS:
+ (theora_enc_finalize), (theora_enc_sink_setcaps):
+ Add a finalize method to ensure we clean up state even if
+ someone omitted the state change back to NULL.
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
+ (gst_vorbisenc_chain):
+ Free some more leaked bits.
+ * tests/check/pipelines/theoraenc.c: (start_pipeline),
+ (stop_pipeline):
+ Wait for state changes to happen if they're ASYNC.
+ This ought to teach those fancy pants buildbots a lesson.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
- Unref leaked buffer. (Noticed by Ronald)
+2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-05 David I. Lehn <dlehn@users.sourceforge.net>
+ gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstid3tag.c:
+ Add mapping for ID3 International Standard Recording Code
+ tag "TSRC"
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- Sync requires with other checks. >= vs =.
+2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-02-06 Jan Schmidt <thaytan@mad.scientist.com>
+ ext/vorbis/vorbisenc.c: Don't leak tag names.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
+ Don't leak tag names.
- * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_video_getcaps),
- (gst_dvdec_video_link), (gst_dvdec_loop):
- * ext/dv/gstdvdec.h:
- rework the caps negotiation so that dvdec works again instead
- of just segfaulting.
+2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
-=== release 0.7.4 ===
+ Split libgsttag docs into multiple sections.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tags.c:
+ Split libgsttag docs into multiple sections.
-2004-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * NEWS: GStreamer Plugins 0.7.4 "For Great Justice" released
- * configure.ac: changed for release
+ Add libgsttag to the docs.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ Add libgsttag to the docs.
-2004-02-05 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
- * gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in:
- * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
- * pkgconfig/gstreamer-libs-uninstalled.pc.in:
- * pkgconfig/gstreamer-play-uninstalled.pc.in:
- reworked patch by David Lehn to fix libdir and includedir for
- uninstalled libraries
- removed play and gconf from gstreamer-libs since they have their
- own pkgconfig files
+ ext/pango/gsttextoverlay.c: Fix clockoverlay.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
+ (gst_text_overlay_init), (gst_text_overlay_src_event),
+ (gst_text_overlay_collected): Fix clockoverlay.
-2004-02-04 David Schleef <ds@schleef.org>
+2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt): Fix a caps
- memleak.
+ docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
+ Original commit message from CVS:
+ * docs/libs/compiling.sgml:
+ Fix typo: it's pkg-config, not pkg-gconfig
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/tmpl/gstgconf.sgml:
+ There is no libgstgconf in 0.10, remove it
+ from the docs.
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
+
+ docs/libs/tmpl/gstcolorbalance.sgml: Updated.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
+ (gst_text_overlay_src_event), (gst_text_overlay_collected):
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
+ (gst_sub_parse_class_init), (gst_sub_parse_init),
+ (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
+ (parse_mpsub), (parser_state_init), (handle_buffer),
+ (gst_sub_parse_chain), (gst_sub_parse_sink_event),
+ (plugin_init):
+ * gst/subparse/gstsubparse.h: Introduce seeking code.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- use correct GST_TAG_ENCODER tag
+2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
+ Original commit message from CVS:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ Add comment about LANGUAGE tag inconsistency (we want
+ ISO-639-1, but extract three-letter identifiers?)
+ * po/POTFILES.in:
+ Add two translatable files.
- * ext/alsa/gstalsa.c: (gst_alsa_change_state):
- be sure to stop the clock when going to paused
- * sys/oss/gstosssink.c: (gst_osssink_change_state):
- reset number of transmitted when going to ready.
- fixes #132935
+2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-02-05 Charles Schmidt <cschmidt2@emich.edu>
+ gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
+ Original commit message from CVS:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/tag/tags.c:
+ (gst_tag_register_musicbrainz_tags_internal),
+ (gst_tag_register_musicbrainz_tags):
+ Forward-port some tags stuff from the 0.8 branch. This is
+ mostly the addition of musicbrainz tags and their mapping
+ to vorbistags, and a vorbistag mapping of the language tag.
- reviewed by Benjamin Otte
+2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- extract track count (fixes #133410)
+ gst/playback/gstplaybin.c: Fix broken code refactoring.
+ Original commit message from CVS:
+ 2006-02-05 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
+ refactoring.
-2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_caps_nego):
- that should be !=, not == (fixes #132519)
+ Add Dirac typefinding and add dirac format to oggmux.
+ Original commit message from CVS:
+ * ext/ogg/gstoggmux.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ Add Dirac typefinding and add dirac format to oggmux.
-2004-02-04 David Schleef <ds@schleef.org>
+2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
- Make sure set_explicit_caps() is called before adding pad.
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
+ * configure.ac:
+ Improve error message for liboil missingness.
+ Original commit message from CVS:
+ Improve error message for liboil missingness.
-2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- bump nano to 2, first prerelease
- put back AM_PROG_LIBTOOL
-
-2004-02-04 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * testsuite/alsa/Makefile.am:
- these are user test apps, not automatic testsuite tests
-
-2004-02-04 David Schleef <ds@schleef.org>
-
- Convert GST_DEBUG_CAPS() to GST_DEBUG():
- * gst/mpeg1videoparse/gstmp1videoparse.c:
- (mp1videoparse_parse_seq):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- * sys/xvideo/gstxwindow.c: (_gst_xwindow_new):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_sinkconnect),
- (gst_xvideosink_getcaps):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- * testsuite/gst-lint: more tests
-
-2004-02-04 David Schleef <ds@schleef.org>
-
- Replace use of GST_PAD_FORMATS_FUNCTION() and similar macros
- with the code that they would expand to.
- * ext/flac/gstflacdec.c: (gst_flacdec_get_src_formats),
- (gst_flacdec_get_src_query_types),
- (gst_flacdec_get_src_event_masks):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_formats),
- (gst_gnomevfssrc_get_query_types),
- (gst_gnomevfssrc_get_event_mask):
-
-2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
- (gst_sinesrc_dispose):
- fix memleak by properly disposing sinesrc
-
-2004-02-04 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_expose):
- * gst-libs/gst/xoverlay/xoverlay.h: Adding the _expose method to tell
- an overlay to redraw the image because it has been exposed.
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_ximage_put), (gst_ximagesink_expose),
- (gst_ximagesink_xoverlay_init), (gst_ximagesink_init):
- * sys/ximage/ximagesink.h: Implement expose method from XOverlay
- interface
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_xvimage_put), (gst_xvimagesink_expose),
- (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_init):
- * sys/xvimage/xvimagesink.h: Implement expose method from XOverlay
- interface
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_type_find):
- more memleak fixage
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
- * gst/typefind/gsttypefindfunctions.c:
- fix memleaks shown by gst-typefind
-
-2004-02-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * common/glib-gen.mak:
- add hack rule to touch .Plo files
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/tuner/Makefile.am:
- remove glib_root variable
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
- set explicit caps before adding the element, so the autopluggers can
- plug correctly.
- * gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
- (mpeg2_sys_type_find), (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (dv_type_find):
- fix memleaks in typefind functions. gst_type_find_suggest takes a const
- argument.
-
-2004-02-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/colorbalance/colorbalance-marshal.list:
- * gst-libs/gst/colorbalance/colorbalance.c:
- * gst-libs/gst/colorbalance/colorbalance.h:
- * gst-libs/gst/colorbalance/colorbalancemarshal.list:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/mixer/mixer-marshal.list:
- * gst-libs/gst/mixer/mixer.c:
- * gst-libs/gst/mixer/mixer.h:
- * gst-libs/gst/mixer/mixermarshal.list:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/tuner/Makefile.am:
- * gst-libs/gst/tuner/tuner-marshal.list:
- * gst-libs/gst/tuner/tuner.c:
- * gst-libs/gst/tuner/tuner.h:
- * gst-libs/gst/tuner/tunermarshal.list:
- use new glib-gen.mak snippet to clean up Makefile.am
- fix various bugs in Makefile.am's
-
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
- handle chain parsing correctly in the multichain case
- * ext/theora/theoradec.c: (gst_theora_dec_init), (_theora_ilog),
- (theora_dec_from_granulepos), (theora_dec_to_granulepos),
- (theora_dec_src_query), (theora_dec_src_event), (theora_dec_event),
- (theora_dec_chain):
- handle events and queries correctly
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * .cvsignore:
- Ignore generated file _stdint.h.
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * gst-libs/gst/colorbalance/Makefile.am:
- * gst-libs/gst/colorbalance/colorbalance.h:
- * gst-libs/gst/mixer/Makefile.am:
- * gst-libs/gst/mixer/mixer.h:
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/play.h:
- * gst-libs/gst/tuner/Makefile.am:
- * gst-libs/gst/tuner/tuner.h:
- Generate enum type code with glib-mkenums.
- * gst-libs/gst/colorbalance/.cvsignore:
- * gst-libs/gst/mixer/.cvsignore:
- * gst-libs/gst/play/.cvsignore:
- * gst-libs/gst/tuner/.cvsignore:
- Ignore generated files.
-
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
-
- * gst-libs/gst/audio/.cvsignore:
- Ignore generated file.
- * gst-libs/gst/audio/Makefile.am:
- Do not install example filter.
+ gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1):
+ Don't put essential function call into
+ g_return_*() macro, otherwise it'll all be
+ replaced by NOOPs when compiling with
+ G_DISABLE_CHECKS defined.
-2004-02-03 David I. Lehn <dlehn@users.sourceforge.net>
+2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * examples/switch/.cvsignore:
- Ignore generated file.
+ * ChangeLog:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggparse.c:
+ * gst/tcp/gsttcpserversink.c:
+ * sys/v4l/v4lsrc_calls.c:
+ * sys/v4l/v4lsrc_calls.h:
+ Just make it compile with --disable-gst-debug.
+ Original commit message from CVS:
+ Just make it compile with --disable-gst-debug.
+
+2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
+ (gst_alsasink_class_init), (gst_alsasink_init),
+ (gst_alsasink_write), (gst_alsasink_reset):
+ * ext/alsa/gstalsasink.h:
+ Add lock to protect alsa calls.
+ Implement reset to flush samples ASAP, does not work
+ with dmix though.
+
+2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock):
+ Ugh.. getting late I guess...
+
+2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_provide_clock),
+ (gst_base_audio_sink_set_property),
+ (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
+ Don't try to provide a clock when we are not negotiated since
+ we might not be able to make it run.
+
+2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (try_to_link_1):
+ Unlinking two source pads is ... hard.
-2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
- * common/m4/ax_create_stdint_h.m4:
- * configure.ac:
- add AX_CREATE_STDINT_H to get correct type definitions for a52dec in
- _stdint.h.
- * Makefile.am:
- remove generated _stdint.h in DISTCLEANFILES
- * ext/a52dec/gsta52dec.c:
- include _stdint.h for a52dec. (should fix #133064)
-
-2004-02-02 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add),
- (gst_tag_to_vorbis_comments):
- Add replaygain support to vorbistag
-
-2004-02-02 Jeremy Simon <jesimon@libertysurf.fr>
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
- (gst_ffmpeg_caps_to_extradata):
- Fix SVQ3 caps flag properties
- Use glib macro for bytes swap
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/audiofile/gstafsink.c: (gst_afsink_plugin_init):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/sndfile/gstsf.c: (plugin_init):
- * gst/avi/gstavi.c: (plugin_init):
- * sys/dxr3/dxr3init.c: (plugin_init):
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l2/gstv4l2.c: (plugin_init):
- remove textdomain calls
- * po/nl.po:
- update Dutch translation
-
-2004-02-02 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
- (gst_play_set_audio_sink): Moving volume in the audio thread for
- instantaneous volume change. Maybe i will add another volume in front
- of visualization later, not sure yet though.
-
-2004-02-02 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
- (gst_ximagesink_handle_xevents): Better X events handling, only take
- the latest events for configure and motion.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents): same.
-
-2004-02-02 Jon Trowbridge <trow@gnu.org>
-
- reviewed by: David Schleef <ds@schleef.org>
-
- Fix memory leaks:
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_register):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_register):
-
-2004-02-02 David Schleef <ds@schleef.org>
-
- code cleanup. Change bzero() to memset(). Remove duplicate ; at ends
- of lines.
- * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_event):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/xvid/gstxviddec.c: (gst_xviddec_src_link):
- * gst-libs/gst/play/play.c: (gst_play_get_sink_element):
- * gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
- * gst/effectv/gstedge.c: (gst_edgetv_sinkconnect):
- * gst/effectv/gstvertigo.c: (gst_vertigotv_sinkconnect):
- * gst/intfloat/float22int.c: (gst_float2_2_int_getcaps),
- (gst_float2_2_int_link):
- * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
- * gst/rtjpeg/RTjpeg.c: (RTjpeg_init_mcompress):
- * gst/tcp/gsttcpsink.c: (gst_tcpsink_init_send):
- * gst/tcp/gsttcpsrc.c: (gst_tcpsrc_init_receive):
- * gst/udp/gstudpsink.c: (gst_udpsink_init_send):
- * gst/udp/gstudpsrc.c: (gst_udpsrc_init_receive):
- * sys/v4l/gstv4lelement.c: (gst_v4lelement_init):
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
- * testsuite/gst-lint: Add tests for bzero and ;;
-
-2004-02-02 David Schleef <ds@schleef.org>
-
- * gst/debug/efence.c: Add fallback if MAP_ANONYMOUS isn't defined.
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/aalib/gstaasink.c: (gst_aasink_open):
- * ext/alsa/gstalsa.c: (gst_alsa_link), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
- (gst_afsink_close_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
- (gst_afsrc_close_file):
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_user_op), (dvdnavsrc_get):
- * ext/esd/esdmon.c: (gst_esdmon_get):
- * ext/esd/esdsink.c: (gst_esdsink_chain), (gst_esdsink_open_audio):
- * ext/faac/gstfaac.c: (gst_faac_chain):
- * ext/faad/gstfaad.c: (gst_faad_chain):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_error_callback),
- (gst_flacdec_loop):
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (gst_gnomevfssrc_open_file):
- * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
- * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_chain):
- * ext/lcs/gstcolorspace.c: (gst_colorspace_srcconnect_func):
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_do_typefind), (gst_id3_tag_chain):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain),
- (gst_ogg_demux_push):
- * ext/raw1394/gstdv1394src.c:
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_lock),
- (gst_sdlvideosink_initsdl), (gst_sdlvideosink_create):
- * ext/sndfile/gstsf.c: (gst_sf_open_file), (gst_sf_close_file),
- (gst_sf_loop):
- * ext/speex/gstspeexenc.c: (gst_speexenc_chain):
- * ext/swfdec/gstswfdec.c: (gst_swfdec_loop):
- * ext/tarkin/gsttarkindec.c: (gst_tarkindec_chain):
- * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_loop):
- * ext/xvid/gstxviddec.c: (gst_xviddec_setup), (gst_xviddec_chain):
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_setup), (gst_xvidenc_chain):
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_element_data), (gst_riff_read_seek),
- (gst_riff_peek_list), (gst_riff_read_list), (gst_riff_read_header):
- * gst/adder/gstadder.c: (gst_adder_loop):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_segment),
- (gst_asf_demux_process_stream), (gst_asf_demux_get_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_init),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- * gst/avi/gstavimux.c: (gst_avimux_stop_file):
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c: (gst_goom_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/intfloat/float22int.c: (gst_float2_2_int_chain):
- * gst/intfloat/gstfloat2int.c: (gst_float2int_loop):
- * gst/intfloat/gstint2float.c: (gst_int2float_chain_gint16):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_uint), (gst_ebml_read_sint),
- (gst_ebml_read_float), (gst_ebml_read_header):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
- (gst_matroska_demux_parse_blockgroup):
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- * gst/oneton/gstoneton.c: (gst_oneton_chain):
- * gst/silence/gstsilence.c: (gst_silence_get):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/smpte/gstsmpte.c: (gst_smpte_loop):
- * gst/speed/gstspeed.c: (speed_loop):
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * gst/volenv/gstvolenv.c: (gst_volenv_chain):
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_loop):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_open),
- (dxr3audiosink_set_mode_pcm), (dxr3audiosink_set_mode_ac3),
- (dxr3audiosink_close):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_open), (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_open),
- (dxr3videosink_close), (dxr3videosink_write_data):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/oss/gstosssink.c: (gst_osssink_chain):
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_buffer_free):
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay),
- (gst_v4l_set_window), (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_get_capabilities), (gst_v4l_open),
- (gst_v4l_set_chan_norm), (gst_v4l_get_signal),
- (gst_v4l_get_frequency), (gst_v4l_set_frequency),
- (gst_v4l_get_picture), (gst_v4l_set_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_capture_init),
- (gst_v4lsrc_requeue_frame), (gst_v4lsrc_try_palette):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get):
- * sys/v4l2/v4l2-overlay_calls.c: (gst_v4l2_set_display),
- (gst_v4l2_set_window), (gst_v4l2_enable_overlay):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_get_norm),
- (gst_v4l2_set_norm), (gst_v4l2_get_input), (gst_v4l2_set_input),
- (gst_v4l2_get_output), (gst_v4l2_set_output),
- (gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
- (gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
- (gst_v4l2_set_attribute):
- * sys/v4l2/v4l2_calls.h:
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
- (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
- (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
- (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
- (gst_v4l2src_capture_stop):
- * sys/vcd/vcdsrc.c: (vcdsrc_open_file):
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_buffer_new),
- (gst_xvideosink_sinkconnect), (gst_xvideosink_chain),
- (gst_xvideosink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain):
-
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/volume/gstvolume.c: (gst_volume_set_volume),
- (gst_volume_get_volume), (volume_class_init), (volume_init),
- (volume_chain_int16), (volume_update_volume):
- * gst/volume/gstvolume.h:
- make code more readable by removing magic numbers
- make mixer interface export 0-100 range
- make it internally map to 0.0-1.0 range so users don't distort
- output by putting the sliders at full volume
+ gst-libs/gst/audio/TODO: Updated.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/TODO:
+ Updated.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
+ On EOS, wait till the last sample is played before posting EOS.
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/play/play.c: (gst_play_tick_callback),
- (gst_play_state_change), (gst_play_seek_to_time):
- block the tick callback for 0.5 secs after doing a seek
+ * tests/check/pipelines/theoraenc.c:
+ comment on my understanding
+ Original commit message from CVS:
+ comment on my understanding
-2004-02-02 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/play/play.c: (gst_play_new):
- check for GError
+ * common:
+ * tests/check/pipelines/theoraenc.c:
+ reformat to fit 80 chars
+ Original commit message from CVS:
+ reformat to fit 80 chars
-2004-02-01 Julien MOUTTE <julien@moutte.net>
+2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
- * gst-libs/gst/play/play.c: (gst_play_seek_to_time),
- (gst_play_new): Accepting NULL GError, blocking time tick while seeking.
- * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
- (gst_ximagesink_chain), (gst_ximagesink_init): s/sinkconnect/sink_link
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
- (gst_xvimagesink_chain), (gst_xvimagesink_init): s/sinkconnect/sink_link
+ gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
+ Original commit message from CVS:
+ 2006-02-01 Philippe Kalaf <burger at speedy dot org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
+ setting queue_delay to zero. Also avoid thread being started if
+ queue_delay is zero.
-2004-02-01 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
- * configure.ac:
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- check for a function added in vorbis 1.1
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start),
- (gst_alsa_drain_audio), (gst_alsa_stop_audio):
- really start/stop clock only on PLAYING <=> PAUSED
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- remove \n from debugging lines
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain):
- make it work when seeking does not
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- reset on DISCONT
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsa.c: (gst_alsa_change_state), (gst_alsa_start):
- start clock on PAUSED=>PLAYING, not later
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- extract correct time for different discont formats
- (gst_alsa_sink_get_time):
- don't segfault when no format is negotiated yet, just return 0
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_handle_event), (gst_ogg_demux_push),
- (gst_ogg_pad_push):
- handle flush and discont events correctly
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event), (vorbis_dec_chain):
- handle discont events correctly
-
-2004-01-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.c: (gst_play_error_quark),
- (gst_play_error_create), (gst_play_error_plugin),
- (gst_play_pipeline_setup), (gst_play_init), (gst_play_new):
- * gst-libs/gst/play/play.h:
- add error handling during creation
- * examples/gstplay/player.c: (main):
- use new gst_play_new
-
-
-2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/theora/theoradec.c: (theora_dec_chain):
- make comments work
- * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
- (vorbis_dec_src_event), (vorbis_dec_chain):
- add encoder tag, fix tag reading to be more error tolerant, change
- BITRATE to NOMINAL_BITRATE, add debugging, don't unref events after
- gst_pad_event_default.
- * gst/tags/gstvorbistag.c:
- (gst_tag_list_from_vorbiscomment_buffer):
- undefine function specific define at end of function
-
-2004-01-31 Jeremy Simon <jesimon@libertysurf.fr>
-
- * ext/flac/gstflac.c: (plugin_init):
- * ext/flac/gstflacdec.c: (gst_flacdec_class_init):
- * ext/flac/gstflacdec.h:
- * ext/flac/gstflacenc.h:
- Fix typos
-
-2004-01-30 David I. Lehn <dlehn@users.sourceforge.net>
-
- * examples/gstplay/player.c: s/gstplay.h/play.h/
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/Makefile.am:
- * gst-libs/gst/play/gstplay.c:
- * gst-libs/gst/play/gstplay.h:
- * gst-libs/gst/play/play.c:
- more surgery, operation complete
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/play/play.old.c:
- * gst-libs/gst/play/play.old.h:
- after CVS surgery by moving, remove
- * gst-libs/gst/play/playpipelines.c:
- remove
-
- * gst/intfloat/float22int.c: (gst_float2_2_int_chain):
- add negotiation error
-
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_event),
- (gst_ogg_demux_push):
- add some seeking debug info
- send a flush when seeking
-
-2004-01-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
+ Original commit message from CVS:
+ * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
+ Make test work again by connecting fakesinks to each decoded pad,
+ which makes the pipeline wait until each fakesink has a buffer
+ queued before going to PAUSED state. At that point we know the
+ decodebin pads are negotiated.
- * configure.ac:
- use AC_C_INLINE
- * configure.ac:
- * ext/Makefile.am:
- * ext/theora/Makefile.am:
- * ext/theora/theoradec.c:
- add theora video decoder. Does just do simple decoding for now and
- has been tested against Theora cvs only. It only works when theora
- is compiled with --enable-static.
- * ext/vorbis/vorbisdec.c: (vorbis_dec_event):
- always reset packetno on DISCONT
+2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/: Pass unhandled queries to the parent class's query function.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
+ (gst_cdda_base_src_handle_event):
+ * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
+ Pass unhandled queries to the parent class's query function.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- Fix audio.
+2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
+ (gst_ogg_pad_src_query):
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
+ * ext/theora/theoradec.c: (theora_dec_src_query),
+ (theora_dec_sink_query):
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
+ (vorbis_dec_sink_query):
+ * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
+ (gst_vorbisenc_sink_query):
+ * gst/adder/gstadder.c: (gst_adder_query):
+ Pass unhandled queries upstream instead of just
+ dropping them (#326447). Also, fix supported
+ query types list for some elements.
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegaudioparse/gstmpegaudioparse.c:
- (mp3_type_frame_length_from_header):
- Fix header parsing - stolen from ffmpeg (thank you! :) ).
+ gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (au_type_find),
+ (paris_type_find), (ilbc_type_find), (plugin_init):
+ Fix typefinding for audio/x-au, audio/x-paris and
+ audio/iLBC-sh. We cannot use the START_WITH macros
+ here, because there can only be one typefind factory
+ with the same name (caps), so the second one would
+ replace the first one and the first one would never
+ be called when doing typefinding (see #161712).
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
+ (vorbis_handle_header_packet), (vorbis_dec_push),
+ (vorbis_handle_data_packet):
+ Use scale_int when we can, add some more scaling.
+ Check packettype before parsing it.
+
+2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Call right _scale functions.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (_theora_granule_time),
+ (theora_dec_src_convert), (theora_dec_sink_convert):
+ Call right _scale functions.
+ Use parameter instead of some other random value.
+
+2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/theora/theoradec.c: Use higher precision timestamps calculation.
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (_theora_granule_frame),
+ (_theora_granule_time), (_inc_granulepos),
+ (theora_dec_src_convert), (theora_dec_sink_convert),
+ (theora_handle_type_packet), (theora_handle_data_packet),
+ (theora_dec_chain):
+ Use higher precision timestamps calculation.
+ Convert some other conversions to _scale.
+
+2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/: initialize gst_controller before using
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_create_sine_table), (plugin_init):
+ * gst/volume/gstvolume.c: (plugin_init):
+ initialize gst_controller before using
- * ext/esd/esdsink.c: (gst_esdsink_init):
- Since we have static pad template caps, we don't need to negotiate;
- either the core errors out or we know the format.
+2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
+ Original commit message from CVS:
+ * tests/check/pipelines/theoraenc.c:
+ * tests/check/pipelines/vorbisenc.c:
+ Define constant using G_GINT64_CONSTANT to avoid errors when
+ passing it around - otherwise it gets truncated to 32 bits.
+ Fixes failing tests.
+
+2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
+
+ sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
+ Original commit message from CVS:
+ 2006-01-31 Andy Wingo <wingo@pobox.com>
+ * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
+ caps being set doesn't have a framerate value. Basically a stopgap
+ measure.
+ * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
+ technically correct enough to put into core though.
+ (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
+ DURATION. Fixes theoraenc ! oggmux.
+ * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
+ fraction, not double.
+
+2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * gst-plugins-base.spec.in:
+ update with latest files
+ Original commit message from CVS:
+ update with latest files
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_seek):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_seek):
- Fix event handling.
+2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-01-30 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ win32/vs7: add vs7 project files created by Sergey Scobich
+ Original commit message from CVS:
+ * win32/vs7:
+ add vs7 project files created by Sergey Scobich
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps):
- removee video/x-theora from vp3 decoder, it doesn't handle raw
- theora streams
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init):
- fix bug with finalizing element that never went to PAUSED
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
- length and position queries were swapped
- * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
- (vorbis_dec_from_granulepos), (vorbis_dec_src_query),
- (vorbis_dec_src_event):
- implement querying time and bytes
+2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
-2004-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
+ win32/vs8: add vs8 project files created by Sergey Scobich
+ Original commit message from CVS:
+ * win32/vs8:
+ add vs8 project files created by Sergey Scobich
- * just about every source file:
- gst_element_error -> GST_ELEMENT_ERROR
+2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
-2004-01-29 Julien MOUTTE <julien@moutte.net>
+ ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
+ timestamp + duration, not just timestamp -- ogg pages should be
+ ordered by stop time. Necessary fix given the change in vorbis
+ timestamps.
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get): Fixing seeking
- emiting FLUSH and even before DISCONT.
- * gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): Fix seeking to
- get the best instant seeking as possible yay!
+2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
-2004-01-29 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * ChangeLog:
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c:
+ * tests/check/pipelines/theoraenc.c:
+ ext/theora/theoraenc.c (theora_enc_sink_setcaps)
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
+ (gst_theora_enc_init): Pull the granule shift out of the encoder.
+ (granulepos_add): New function, handles the messiness of adjusting
+ granulepos values.
+ (theora_buffer_from_packet):
+ (theora_enc_chain):
+ (theora_enc_sink_event): Use granulepos_add, not +.
+ * tests/check/pipelines/theoraenc.c
+ (check_buffer_granulepos_from_starttime): Just check the frame
+ count, not the actual granulepos -- we can't dictate to the
+ encoder when it should be placing keyframes.
+
+2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
+ SERVICE_NOT_AVAILABLE happens for example when you're trying to
+ play an http:// stream from a server that's not serving
- * gst/mpeg1videoparse/gstmp1videoparse.c:
- (gst_mp1videoparse_real_chain):
- Committed wrong version last week... Grr... Didn't notice until now.
+2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
+ * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
+ remove the UINT64_CONSTANT macro, doesn't appear to be needed or
+ available.
+
+2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
+
+ ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * ext/theora/gsttheoraenc.h:
+ * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
+ although theoraenc was timestamping correctly. Added handling of
+ streams that start with nonzero timestamps.
+ * tests/check/Makefile.am:
+ * tests/check/pipelines/theoraenc.c: New file, basically does same
+ tests as vorbisenc.
+ * tests/check/pipelines/vorbisenc.c: I claim these bugs.
+
+2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
+ (gst_audioringbuffer_pause):
+ Implement pause that does not wait for completion.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Don't drop buffers when going to PAUSED but perform preroll on
+ remaining samples now that core base class supports this.
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
+ (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
+ (gst_ring_buffer_commit):
+ Pause should not signal waiters.
+ Implement return value of _commit correctly.
+
+2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
+
+ tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
+ Original commit message from CVS:
+ 2006-01-30 Andy Wingo <wingo@pobox.com>
+ * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
+ * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
+ updated to timestamp from the first sample, not the last.
+ (gst_vorbisenc_buffer_from_header_packet): New function, takes
+ special care of granulepos and timestamp for header packets.
+ (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
+ when the first buffer has a nonzero timestamp.
+ * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
+ (GstVorbisEnc.subgranule_offset): New members. Take care of the
+ case when the first audio buffer we get has a nonzero timestamp.
+ (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
+ properly timestamp vorbis buffers with the time of the first
+ sample, not the last.
+ * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
+ vorbis_granule_time_copy -- now it takes the granule/subgranule
+ offset into account.
+ * tests/check/pipelines/vorbisenc.c: New test for correctness of
+ timestamps, durations, and granulepos on buffers produced by
+ vorbisenc.
+
+2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
+
+ gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
+ Original commit message from CVS:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
+ Patch from Eric Jonas to support conversions to/from UYVY
+ (Fixes: #324626)
-2004-01-29 Julien MOUTTE <julien@moutte.net>
+2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/: Implement subtitles.
+ Original commit message from CVS:
+ 2006-01-30 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybasebin.c: (group_commit),
+ (queue_overrun),
+ (setup_subtitle), (setup_source), (set_active_source):
+ * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
+ (gen_text_element), (gen_audio_element), (gen_vis_element),
+ (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
+
+2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
+
+ gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
+ Original commit message from CVS:
+ * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
+ * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
+ use of gst_guint64_to_gdouble to be compliant with vs6
+ * gst/playback/gstdecodebin.c: (try_to_link_1)
+ * gst/videorate/videorate.c: (gst_video_rate_blank_data)
+ use of G_GINT64_CONSTANT for int64 constants
+ * win32/common/libgstinterfaces.def:
+ export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
+ * win32/vs6:
+ update and add new project files
+
+2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new): Emit the
- have_xwindow_id signal in xwindow_create.
+ add a win32-update rule like in core, and copy over enumtypes files
+ Original commit message from CVS:
+ * Makefile.am:
+ * win32/MANIFEST:
+ * win32/common/interfaces-enumtypes.c:
+ (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
+ (gst_mixer_track_flags_get_type),
+ (gst_tuner_channel_flags_get_type):
+ * win32/common/interfaces-enumtypes.h:
+ * win32/common/multichannel-enumtypes.c:
+ (gst_audio_channel_position_get_type):
+ * win32/common/multichannel-enumtypes.h:
+ add a win32-update rule like in core, and copy over enumtypes files
-2004-01-29 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/ogg/gstoggdemux.c:
- lots of changes - mainly support for chained bitstreams, seeking,
- querying and bugfixes of course
- * ext/vorbis/Makefile.am:
- * ext/vorbis/vorbisdec.c:
- * ext/vorbis/vorbisdec.h:
- add vorbisdec raw vorbis decoder
- * ext/vorbis/vorbis.c: (plugin_init):
- register vorbisdec as PRIMARY, vorbisfile as SECONDARY
- * gst/intfloat/Makefile.am:
- * gst/intfloat/float22int.c:
- * gst/intfloat/float22int.h:
- * gst/intfloat/gstintfloatconvert.c: (plugin_init):
- add float2intnew plugin. It converts multichannel interleaved float to
- multichannel interleaved int. The name should probably be changed.
- * gst/typefind/gsttypefindfunctions.c: (theora_type_find),
- (plugin_init):
- add typefinding for raw theora video so oggdemux can detect it.
-
-2004-01-28 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): seek on video
- sink element first.
- * gst/videoscale/gstvideoscale.c:
- (gst_videoscale_handle_src_event): Fixing src event handler.
-
-2004-01-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
- (gst_v4lsrc_open), (gst_v4lsrc_close),
- (gst_v4lsrc_palette_to_caps), (gst_v4lsrc_srcconnect),
- (gst_v4lsrc_getcaps), (gst_v4lsrc_set_clock):
- * sys/v4l/gstv4lsrc.h:
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_start),
- (gst_v4lsrc_grab_frame), (gst_v4lsrc_capture_stop):
- Implement resizing... Hack. But that's why v4l is b0rked...
+ * configure.ac:
+ generate win32/common/config.h
+ Original commit message from CVS:
+ generate win32/common/config.h
-2004-01-28 Julien MOUTTE <julien@moutte.net>
+2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
- (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_ximage_put), (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_destroy),
- (gst_xvimagesink_xwindow_new), (gst_xvimagesink_xwindow_destroy),
- (gst_xvimagesink_xwindow_resize), (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_xcontext_get): Removing some useless debugs messages,
- correctly cleaning the image created to check xshm calls on succes,
- added a lot of XSync calls in X11 functions, and fixed a segfault when
- no image format was defined before negotiation happened.
+ win32/: add config files just like in core
+ Original commit message from CVS:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ * win32/common/config.h.in:
+ add config files just like in core
-2004-01-28 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/alsa/gstalsa.c: (gst_alsa_query_func):
- use gst_element_get_time to get correct time
+ ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
+ (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
+ (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
+ * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
+ (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
+ (gst_alsasrc_unprepare), (gst_alsasrc_read):
+ Update all error messages. All of them should either use
+ the default translated message, or actually provide a
+ translatable string.
+ Make the string for channel count problems meaningful.
-2004-01-28 Julien MOUTTE <julien@moutte.net>
+2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xerror),
- (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_destroy),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_class_init):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xerror),
- (gst_xvimagesink_check_xshm_calls),
- (gst_xvimagesink_xvimage_destroy), (gst_xvimagesink_xcontext_get): Our
- X plugins are now able to detect that XShm calls will fail even if the
- server claims that it has XShm support (remote displays most of the
- time). We then log the error as a GST_DEBUG and set use_shm to FALSE
- so that we use non XShm functions. This feature is almost useless for
- xvimagesink as Xv is not supported on remote displays anyway, but
- it might happen than even on the local display XShm calls fail.
+ gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
+ Make gcc-4.1 happy (part of #327357).
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_init),
- (gst_esdsink_link), (gst_esdsink_get_time), (gst_esdsink_chain),
- (gst_esdsink_change_state): Fix sync issues in esdsink. Also
- changed esdsink to only use 44100,16,2, since esd sucks at rate
- conversion and esdsink has had difficulty negotiating.
+ sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
+ Original commit message from CVS:
+ * sys/v4l/v4l_calls.c: (gst_v4l_open):
+ check for and throw RESOURCE_BUSY
-2004-01-27 Julien MOUTTE <julien@moutte.net>
+2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
- * gst-libs/gst/play/gstplay.c: (gst_play_tick_callback),
- (gst_play_seek_to_time): Fixing the way to get current position.
+ gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
+ Original commit message from CVS:
+ * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
+ checked in this change -- it requires liboil features not
+ in 0.3.6. Revert parts.
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
- * sys/oss/gstosssink.c: (gst_osssink_sink_query):
- use gst_element_get_time to get correct time
+ update liboil requirement to 0.3.6
+ Original commit message from CVS:
+ * REQUIREMENTS:
+ * configure.ac: update liboil requirement to 0.3.6
+ * gst/videoscale/Makefile.am:
+ * gst/videoscale/vs_scanline.c: liboilify
-2004-01-27 Julien MOUTTE <julien@moutte.net>
+2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst-libs/gst/play/gstplay.c: (gst_play_set_location): The easiest
- fix ever... Inverting 2 lines of code make spider autoplug correctly
- tagged mp3 !
+ ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (get_buffer):
+ When pad_alloc returns a GstFlowReturn other
+ than GST_FLOW_OK, make sure it is passed upstream.
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Use gst_pad_try_set_caps_nonfixed().
+ ext/alsa/gstalsasink.c: Free the device name string.
+ Original commit message from CVS:
+ * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
+ (gst_alsasink_class_init):
+ Free the device name string.
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
+ (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
+ (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
+ Don't remove a pad from the collectpads structure until it
+ is released - it's a request pad, and may receive data again
+ if the element gets moved back to PLAYING state.
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
+ Ensure we turn on double buffering on the Xv port, and
+ set the colour key to something dark and mysterious that
+ isn't black.
-2004-01-27 David Schleef <ds@schleef.org>
+2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ ext/: - a library should not call setlocale. see Libraries node in gettext manual
+ Original commit message from CVS:
+ * ext/alsa/gstalsaplugin.c: (plugin_init):
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_src_base_init), (plugin_init):
+ * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
+ - a library should not call setlocale. see Libraries node in
+ gettext manual
+ - make sure all plugins that use translation do bindtextdomain
+ to point to the localedir
+ * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
+ (setup_sinks), (plugin_init):
+ all this, and check for NULL when creating sinks
- * gst/ac3parse/gstac3parse.c: update to checklist 5
- * gst/adder/gstadder.c: rewrite negotiation. update to checklist 5
- * gst/audioconvert/gstaudioconvert.c: update to checklist 5
- * gst/audioscale/gstaudioscale.c: same
- * gst/auparse/gstauparse.c: same
- * gst/avi/gstavidemux.c: same
+2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
+ Original commit message from CVS:
+ 2006-01-27 Julien MOUTTE <julien@moutte.net>
+ * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
+ (plugin_init): Make typefinding of subtitles work again.
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_handle_sink_event):
- stop processing after EOS
+2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
+ Original commit message from CVS:
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
+ (mp3_type_frame_length_from_header), (mp3_type_find),
+ (wavpack_type_find), (m4a_type_find), (ircam_type_find),
+ (plugin_init):
+ Backport a bunch of typefinding fixes from the 0.8 branch.
+ Also, improve wavpack typefinding: if we can't peek the
+ entire wavpack block, try to parse the bits we can get and
+ see if we find what we're looking for in those.
- * gst/asfdemux/asfheaders.h:
- * gst/asfdemux/gstasfdemux.c:
- * gst/asfdemux/gstasfmux.c: (gst_asfmux_put_guid),
- (gst_asfmux_put_string), (gst_asfmux_put_wav_header),
- (gst_asfmux_put_vid_header), (gst_asfmux_put_bmp_header):
- lot's of fixes to make data extraction simpler and get the code
- architecture and compiler independant. Add debugging category
- * gst/goom/gstgoom.c: (gst_goom_change_state):
- reset channel count on PAUSED=>READY, not READY=>PAUSED
+2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Handle some more cases of pixel aspect ratio.
+ Original commit message from CVS:
+ 2006-01-26 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c:
+ (gst_ximagesink_calculate_pixel_aspect_ratio):
+ * sys/xvimage/xvimagesink.c:
+ (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
+ more cases of pixel aspect ratio.
+
+2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (pad_probe):
+ Also consider the flush-start and tag events as unblockers
+ for the pad probes.
+
+2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
+
+ gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
+ Original commit message from CVS:
+ 2006-01-26 Julien MOUTTE <julien@moutte.net>
+ * gst/playback/gstplaybin.c: (gst_play_bin_init),
+ (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
+ (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
+ On the fly visualisation switch, works disabling, enabling as
+ well but it won't be able to enable vis in a playbin that was
+ created with no visualisation.
+
+2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
+ Undo previous commit, it breaks resume after pause.
+
+2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
+ (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
+ Improve debugging.
+ Post error when caps cannot be parsed.
+ Resync on discontinuity in the stream.
+ Clip samples to segment boundaries.
+ return WRONG_STATE sooner when we are flushing.
+ * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
+ (gst_base_audio_src_get_time), (gst_base_audio_src_create):
+ Make audiosrc operate in TIME.
+ Set TIMESTAMP and DURATION on buffers.
+
+2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ tests/examples/seek/seek.c: Output tag messages as well.
+ Original commit message from CVS:
+ * tests/examples/seek/seek.c: (main):
+ Output tag messages as well.
-2004-01-26 Colin Walters <walters@verbum.org>
+2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
+ (free_pad_probes), (remove_fakesink), (pad_probe),
+ (close_pad_link), (gst_decode_bin_change_state):
+ Replace GstPadBlockCallback with pad probes that detect
+ first buffer AND eos before removing fakesink.
+ Fixes hang with demuxers doing EOS while pre-rolling.
+ Solves #328279
+
+2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
+
+ GCC 2.95 fixes (#328263).
+ Original commit message from CVS:
+ 2006-01-23 Andy Wingo <wingo@pobox.com>
+ * ext/alsa/gstalsasink.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_setcaps),
+ (gst_base_rtp_depayload_add_to_queue),
+ (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
+ Patch by: Jens Granseuer <jensgr at gmx dot net>
+
+2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
+
+ sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
+ Original commit message from CVS:
+ 2006-01-22 Julien MOUTTE <julien@moutte.net>
+ * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
+ (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
+ frames. We might get a frame destroyed after changing state to
+ NULL, adding a safety check on xcontext.
+
+2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
+ Original commit message from CVS:
+ * gst-libs/gst/interfaces/xoverlay.c:
+ Fix prepare-xwindow-id code example in the docs - we need to
+ ignore all messages that aren't element messages as well.
- * ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_get): Remove ugly
- code to pull a bigger buffer in iradio mode. This as a side effect
- makes typefinding work.
+2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
+ sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
+ Original commit message from CVS:
+ 2006-01-21 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
+ I think one day i'll completely undestand how caps negotiation
+ is supposed to work. This refactoring handles buffer_alloc
+ called with caps we can't handle. We definitely don't want a
+ set_caps with those caps, so we define and allocate a buffer
+ we would like to receive.
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_extradata):
- Fix SVQ3 decoding on PPC
+2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-26 Julien MOUTTE <julien@moutte.net>
+ * autogen.sh:
+ * common:
+ up automake requirement to 1.7
+ Original commit message from CVS:
+ up automake requirement to 1.7
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Dunno how
- that one managed to stay there... Fixed.
+2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-26 Jeremy Simon <jesimon@libertysurf.fr>
+ gst/playback/gstplaybasebin.c: Free iterator when done.
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Free iterator when done.
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
- (gst_ffmpeg_caps_to_extradata), (gst_ffmpeg_caps_to_pixfmt):
- * gst/qtdemux/qtdemux.c: (plugin_init), (qtdemux_parse_trak),
- (qtdemux_video_caps):
- * gst/qtdemux/qtdemux.h:
- Add SVQ3 specific flags to qtdemux and ffmpeg
+2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_render):
+ Fix playback of non-synchronised streams by assuming a rate
+ of 1.0 instead of a random one.
+ Makes this work again:
+ gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
+ endianness=(int)4321, signed=(boolean)true, width=(int)16,
+ depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
+ audioresample ! alsasink
- * gst-libs/gst/audio/audio.h:
- remove buffer-frames from audio caps
- * gst/audioconvert/gstaudioconvert.c:
- fix plugin to really work.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/mixer/mixer.c:
- * gst-libs/gst/propertyprobe/propertyprobe.c:
- * gst-libs/gst/tuner/tuner.c: (gst_tuner_find_norm_by_name),
- (gst_tuner_find_channel_by_name):
- * gst-libs/gst/tuner/tuner.h:
- Add gtk-doc style comments. Also fix a function name.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c: (gst_divxdec_init),
- (gst_divxdec_negotiate):
- Fix for new capsnego - also fixes gst-player with divxdec.
-
-2004-01-25 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_identity_handoff), (gst_play_set_location),
- (gst_play_set_visualization), (gst_play_connect_visualization): Another
- try in visualization implementation. Still have an issue with switch
- blocking when pulling from video_queue and only audio comes out of
- spider.
- * gst/switch/gstswitch.c: (gst_switch_release_pad),
- (gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
- release method. And check if the pad is usable before pulling.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose),
- (gst_videobalance_init),
- (gst_videobalance_colorbalance_list_channels),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value),
- (gst_videobalance_update_properties),
- (gst_videobalance_update_tables_planar411),
- (gst_videobalance_planar411):
- * gst/videofilter/gstvideobalance.h:
- Implement lookup-tables. +/- 10x faster.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
- The index reading was broken. The rest worked fine, but the whole
- goal of my rewrite was to make avidemux readable, and this was
- not at all readable. Please use typed variables.
-
-2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- Additional pad usability check.
- * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
- (mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
- (gst_mp1videoparse_real_chain):
- Fix MPEG video stream parsing. The original plugin had several
- issues, including not timestamping streams where the source was
- not timestamped (this happens with PTS values in mpeg system
- streams, but MPEG video is also a valid stream on its own so
- that needs timestamps too). We use the display time code for that
- for now. Also, if one incoming buffer contains multiple valid
- frames, we push them all on correctly now, including proper EOS
- handling. Lastly, several potential segfaults were fixed, and we
- properly sync on new sequence/gop headers to include them in next,
- not previous frames (since they're header for the next frame, not
- the previous). Also see #119206.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
- (bpf_from_header):
- Move caps setting so we only do it after finding several valid
- MPEG-1 fraes sequentially, not right after the first one (which
- might be coincidental).
- * gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
- (mpeg_video_type_find), (mpeg_video_stream_type_find),
- (plugin_init):
- Add unsynced MPEG video stream typefinding, and change some
- probability values so we detect streams rightly. The idea is as
- follows: I can have an unsynced system stream which contains
- video. In the current code, I would randomly get a type for either
- system or video stream type found, because the probabilities are
- being calculated rather randomly. I now use fixed values, so we
- always prefer system stream if that was found (and that is how it
- should be). If no system stream was found, we can still identity the stream as video-only.
-
-2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
- (gst_avi_demux_stream_odml), (gst_avi_demux_stream_index):
- don't write to buffer. Extract data without the need of
- __attribute__ ((packed))
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (mpeg1_parse_header),
- (mpeg1_sys_type_find):
- Fix MPEG-1 stream typefinding.
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
- Fix typefinding for MPEG-1 system streams, similar to MPEG-2.
-
-2004-01-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/esd/esdsink.c: (gst_esdsink_open_audio):
- * ext/esd/gstesd.c: (plugin_init):
- private debugging, better error reporting
-
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
- (gst_riff_read_init), (gst_riff_read_change_state):
- * gst-libs/gst/riff/riff-read.h:
- Remove stuff fromold metadata system.
+2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * ext/ogg/gstoggdemux.c:
- Fix wrong file comment.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
- Add metadata reading properly.
+=== release 0.10.2 ===
-2004-01-23 Thomas Vander Stichele <thomas at apestaart dot org>
+2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/Makefile.am:
- Fix nas DIST_SUBDIRS
- Uraeus:
- Fix bug where make distcheck doesn't get run on adding stuff to
- the build.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.2
+ Original commit message from CVS:
+ releasing 0.10.2
-2004-01-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
- * ext/divx/gstdivxdec.c: (gst_divxdec_init), (gst_divxdec_setup):
- * ext/divx/gstdivxdec.h:
- Fix divx3 ("msmpeg4") playback using divxdec.
+ gst/playback/: Comment out broken code that connects to the state-changed signal.
+ Original commit message from CVS:
+ * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_set_property):
+ Comment out broken code that connects to the state-changed signal.
+ At this point, changing current stream selection is broken, but
+ stuff like gst-launch playbin current-audio=1 works and filters
+ to the chosen stream.
-2004-01-23 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/typefind/gsttypefindfunctions.c:
- (mp3_type_frame_length_from_header): fix bug in length computation
- (mp3_type_find): improve debugging output
-
-2004-01-23 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_set_location), (gst_play_seek_to_time),
- (gst_play_set_audio_sink), (gst_play_set_visualization),
- (gst_play_connect_visualization), (gst_play_get_sink_element): Reworked
- the pipeline from scratch. Visualization is back and switch went out as
- i realized it was not possible to use the way i wanted.
- * sys/ximage/ximagesink.c: (gst_ximagesink_imagepool_clear),
- (gst_ximagesink_change_state), (gst_ximagesink_dispose): Move xcontext
- clearing in state change from READY to NULL. So that one can clean the
- X ressources keeping the element.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_imagepool_clear), (gst_xvimagesink_change_state),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_set_property), (gst_xvimagesink_dispose),
- (gst_xvimagesink_init): Same xcontext cleaning than ximagesink in state
- change from READY to NULL and fixed some stupid bugs in colorbalance
- get/set values. Also added the following feature : when nobody tries to
- set some values to the colorbalance levels before the xcontext is
- grabbed, then when creating channels list from Xv attributes we set the
- internal values to the Xv defaults. This way we handle buggy Xv drivers
- that set default hue values far from the middle of the range (Thanks
- to Jon Trowbridge for pointing that issue).
- * sys/xvimage/xvimagesink.h: Adding a cb_changed boolean to know if
- colorbalance levels have been set before xcontext is grabbed.
-
-2004-01-22 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
- Fix the ossmixer case where we shouldn't open /dev/dsp* because
- it might block operations (which is bad for a mixer).
-
-2004-01-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/media-info/media-info-priv.c: (have_type_callback),
- (deep_notify_callback), (gmi_set_decoder), (gmi_clear_decoder),
- (gmip_find_type_pre):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info.c:
- (gst_media_info_instance_init), (gst_media_info_read_idler):
- add fakesink to get caps on decoder src pad again
- fix callback prototype to match new have_type signal signature
-
-2004-01-22 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/adder/gstadder.c: (gst_adder_link):
- fix non-compile and cut-n-paste code
-
-2004-01-21 David Schleef <ds@schleef.org>
-
- * ext/swfdec/gstswfdec.c: (gst_swfdec_video_getcaps),
- (gst_swfdec_video_link), (copy_image), (gst_swfdec_loop),
- (gst_swfdec_init), (gst_swfdec_change_state):
- * ext/swfdec/gstswfdec.h:
- Fix negotiation.
- * gst/adder/gstadder.c: (gst_adder_link), (gst_adder_init),
- (gst_adder_request_new_pad): Fix negotiation.
- * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_fixate):
- Add a fixate function.
- * gst/intfloat/gstfloat2int.c:
- * gst/intfloat/gstfloat2int.h:
- * gst/intfloat/gstint2float.c:
- * gst/intfloat/gstint2float.h:
- Completely rewrite the negotiation. Doesn't quite work yet,
- due to some buffer-frames problem.
-
-2004-01-21 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
+ Original commit message from CVS:
+ * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
+ Fix #327216 (null dereference in vorbisdec)
- * ext/gnomevfs/gstgnomevfssrc.c:
- * sys/v4l2/v4l2_calls.h:
- fix includes for distcheck
+2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-21 Christian Schaller <uraeus@gnome.org>
+ ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
+ Original commit message from CVS:
+ * ext/theora/theoradec.c: (theora_handle_comment_packet):
+ Post taglist actually on bus instead of just freeing it
+ (fixes #327114 and totem bug #327080).
+ * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
+ Use gst_element_found_tags_for_pad(), so that the tags
+ are sent downstream as an event as well.
- * ext/nas/
- Add libnas (network audio system) plugin, patch from Arwed von Merkatz
- based on earlier patch from Laurent Vivier
+2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Jeremy Simon <jesimon@libertysurf.fr>
+ sys/: move all regularly occurring messages to GST_LOG level add some more object logs
+ Original commit message from CVS:
+ * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
+ (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
+ (gst_ximagesink_buffer_alloc):
+ * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
+ (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
+ (gst_xvimagesink_buffer_alloc):
+ move all regularly occurring messages to GST_LOG level
+ add some more object logs
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_extradata):
- Fix wma caps property
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_audio_caps):
- Fix typo (flags1 and flags2)
+2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * configure.ac:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ prerelease
+ Original commit message from CVS:
+ prerelease
- * gst-libs/gst/media-info/media-info-priv.c:
- (deep_notify_callback), (gmi_seek_to_track), (gmi_get_decoder),
- (gmip_find_type_pre), (gmip_find_type), (gmip_find_stream_post),
- (gmip_find_stream), (gmip_find_track_metadata),
- (gmip_find_track_streaminfo_post), (gmip_find_track_streaminfo),
- (gmip_find_track_format):
- * gst-libs/gst/media-info/media-info-priv.h:
- * gst-libs/gst/media-info/media-info-test.c: (main):
- * gst-libs/gst/media-info/media-info.c: (gst_media_info_init),
- (gst_media_info_read_idler), (gst_media_info_read):
- * gst-libs/gst/media-info/media-info.h:
- register debugging category and use it for debugging
+2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/ogg/gstoggmux.c: fix a silly segfault
+ Original commit message from CVS:
+ 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
+ fix a silly segfault
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_update_streaminfo),
- (gst_vorbisfile_new_link):
- signal streaminfo through tags
+2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ Add docs for mixerutils stuff.
+ Original commit message from CVS:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/audio/mixerutils.c:
+ * gst-libs/gst/audio/mixerutils.h:
+ Add docs for mixerutils stuff.
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- g++ doesn't like NULL in our i18n/error macros, should be
- either (NULL) or ("").
+2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (setup_source):
+ Fix playback for sources that emit raw audio or
+ raw video streams (e.g.: cd audio sources) (#325984).
- * sys/dxr3/dxr3audiosink.c:
- * sys/dxr3/dxr3init.c:
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_close):
- Fix more error error error errors (missing includes here).
+2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/audio/mixerutils.c: actually save the element we create
+ Original commit message from CVS:
+ * gst-libs/gst/audio/mixerutils.c:
+ (gst_audio_mixer_filter_do_filter):
+ actually save the element we create
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- fix thomas' error errors.
+2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ * gst-plugins-base.spec.in:
+ remove version suffix
+ Original commit message from CVS:
+ remove version suffix
- * ext/mpeg2enc/gstmpeg2enc.cc:
- fix error errors.
+2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_handle_track_seek):
+ No need to post a tag message on the bus when seeking
+ within the same track, only post it when the current
+ track changes.
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- Fix for new error system.
+2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (group_destroy),
+ (probe_triggered), (new_decoded_pad), (mute_group_type),
+ (set_active_source):
+ * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
+ * gst/playback/gststreamselector.c:
+ (gst_stream_selector_base_init),
+ (gst_stream_selector_set_property),
+ (gst_stream_selector_request_new_pad):
+ Reenable stream selection. These mechanisms need a complete overhaul
+ in the face of 0.8->0.10 changes though.
- * ext/divx/gstdivxenc.c: (gst_divxenc_setup), (gst_divxenc_chain):
- fix for new error reporting
+2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-20 David Schleef <ds@schleef.org>
+ ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c:
+ Change the pad template to src_%d to match the pads that
+ are created from it. decodebin needs this information in order
+ to decide that oggdemux is capable of producing multiple pads
+ (and hence needs queues inserted).
+ * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
+ (gst_ogg_mux_collected):
+ Make debug output more useful by using GST_PTR_FORMAT.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_xcontext_get), (gst_ximagesink_getcaps),
- (gst_ximagesink_set_xwindow_id): Change to using a framerate
- of [1,100] instead of [0,MAX], since 0 isn't handled correctly,
- and neither is 100+, most likely.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
- (gst_xvimagesink_getcaps): same
+2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
-2004-01-19 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * gst-plugins-base.spec.in:
+ update spec.in file
+ Original commit message from CVS:
+ update spec.in file
- * configure.ac:
- Up version requirement to 2.0.3 (not yet released) to avoid symbol
- clashes with ffmpeg.
+2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-20 Julien MOUTTE <julien@moutte.net>
+ gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
+ Original commit message from CVS:
+ Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
+ Set depth and width for alaw/mulaw (fixes #326601).
- * gst/switch/gstswitch.c: (gst_switch_request_new_pad),
- (gst_switch_init): Fixed switch element : proxying link and setting
- caps from src to sink on request.
+2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ tests/icles/Makefile.am: don't build the tests if we don't have the libs
+ Original commit message from CVS:
+ * tests/icles/Makefile.am:
+ don't build the tests if we don't have the libs
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_overlay):
- * sys/v4l2/v4l2-overlay_calls.c: (gst_v4l2_set_display),
- (gst_v4l2_set_window), (gst_v4l2_enable_overlay):
- fix element_error
+2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
+ Original commit message from CVS:
+ * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
+ (gst_cd_paranoia_paranoia_callback):
+ Don't try to free NULL pointers.
- * sys/v4l/v4l_calls.h:
- * sys/v4l2/v4l2_calls.h:
- element_error fixes
+2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst/audiorate/gstaudiorate.c: Add debugging category.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
+ (gst_audio_rate_change_state), (plugin_init):
+ Add debugging category.
+ Fix type issues.
+ Add case for incoming buffers without valid offset/offset_end.
- * gst-libs/gst/gst-i18n-plugin.h:
- add locale.h
- remove config.h inclusion
+2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
+ gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
+ Don't leak GCond in audio sources.
- * autogen.sh:
- adding autopoint invocation
- * Makefile.am:
- * configure.ac:
- * gst-libs/gst/gettext.h:
- adding gettext bits
- * ext/audiofile/gstafsink.c: (gst_afsink_plugin_init):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_plugin_init):
- * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_open_file):
- * ext/sndfile/gstsf.c: (gst_sf_loop), (plugin_init):
- * gst-libs/gst/gst-i18n-plugin.h:
- * gst/avi/gstavi.c: (plugin_init):
- * sys/dxr3/dxr3init.c: (plugin_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_write_data):
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/v4l/gstv4l.c: (plugin_init):
- * sys/v4l/v4l_calls.c: (gst_v4l_open):
- * sys/v4l2/gstv4l2.c: (plugin_init):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_get_norm), (gst_v4l2_set_norm),
- (gst_v4l2_get_input), (gst_v4l2_set_input), (gst_v4l2_get_output),
- (gst_v4l2_set_output), (gst_v4l2_get_frequency),
- (gst_v4l2_set_frequency), (gst_v4l2_signal_strength),
- (gst_v4l2_get_attribute), (gst_v4l2_set_attribute):
- make sure locale and translation domain are set
- fix translated strings
- * po/.cvsignore:
- * po/LINGUAS:
- * po/Makevars:
- * po/POTFILES.in:
- * po/nl.po:
- put translation files into place
- * sys/xvideo/imagetest.c: (main):
- * ext/dv/demo-play.c: (main):
- fix unnecessary translations
+2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
-2004-01-19 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+ gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
+ Original commit message from CVS:
+ * gst/playback/gstplaybin.c: (gen_audio_element):
+ Don't leak an autoaudiosink/alsasink when we generate
+ a new audio element. (old code, I guess)
- * ext/sndfile/gstsf.c:
- * gst/avi/gstavimux.c:
- * ext/audiofile/gstafsink.c:
- * ext/audiofile/gstafsrc.c:
- * ext/gnomevfs/gstgnomevfssink.c:
- * ext/gnomevfs/gstgnomevfssrc.c:
- * sys/oss/gstosselement.c:
- * sys/v4l/v4l_calls.h:
- fix i18n include
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
- (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_get_norm),
- (gst_v4l2_set_norm), (gst_v4l2_get_input), (gst_v4l2_set_input),
- (gst_v4l2_get_output), (gst_v4l2_set_output),
- (gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
- (gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
- (gst_v4l2_set_attribute):
- update to new error handling
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ext/sidplay/gstsiddec.cc:
- * gst/modplug/gstmodplug.cc:
- parenthese NULL because C++ seems angry about it
-
-2004-01-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/gst-i18n-plugin.h:
- add skeleton i18n stuff, but needs to be further implemented
-
-2004-01-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * examples/gstplay/player.c: (main):
- * ext/aalib/gstaasink.c: (gst_aasink_open):
- * ext/alsa/gstalsa.c: (gst_alsa_link), (gst_alsa_xrun_recovery):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_loop):
- * ext/audiofile/gstafsink.c: (gst_afsink_open_file),
- (gst_afsink_close_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
- (gst_afsrc_close_file):
- * ext/divx/gstdivxdec.c: (gst_divxdec_setup), (gst_divxdec_chain):
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_user_op), (dvdnavsrc_get):
- * ext/esd/esdmon.c: (gst_esdmon_get):
- * ext/esd/esdsink.c: (gst_esdsink_chain):
- * ext/faac/gstfaac.c: (gst_faac_chain):
- * ext/faad/gstfaad.c: (gst_faad_chain):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_error_callback),
- (gst_flacdec_loop):
- * ext/flac/gstflacenc.c: (gst_flacenc_chain):
- * ext/flac/gstflactag.c: (gst_flac_tag_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
- * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_open_file),
- (gst_gnomevfssink_close_file):
- * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
- (gst_gnomevfssrc_open_file):
- * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop):
- * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_chain):
- * ext/lcs/gstcolorspace.c: (gst_colorspace_srcconnect_func):
- * ext/mad/gstid3tag.c: (gst_id3_tag_handle_event),
- (gst_id3_tag_do_typefind), (gst_id3_tag_chain):
- * ext/mad/gstmad.c: (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_loop):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain),
- (gst_ogg_demux_push), (gst_ogg_pad_push):
- * ext/raw1394/gstdv1394src.c:
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_lock),
- (gst_sdlvideosink_initsdl), (gst_sdlvideosink_create):
- * ext/sidplay/gstsiddec.cc:
- * ext/sndfile/gstsf.c: (gst_sf_open_file), (gst_sf_close_file),
- (gst_sf_loop):
- * ext/speex/gstspeexenc.c: (gst_speexenc_chain):
- * ext/tarkin/gsttarkindec.c: (gst_tarkindec_chain):
- * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_chain):
- * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_loop):
- * ext/xvid/gstxviddec.c: (gst_xviddec_setup), (gst_xviddec_chain):
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_setup), (gst_xvidenc_chain):
- * gst-libs/gst/Makefile.am:
- * gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
- (gst_riff_read_element_data), (gst_riff_read_seek),
- (gst_riff_peek_list), (gst_riff_read_list), (gst_riff_read_header):
- * gst/adder/gstadder.c: (gst_adder_parse_caps), (gst_adder_loop):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_segment),
- (gst_asf_demux_process_stream), (gst_asf_demux_get_stream):
- * gst/avi/gstavidemux.c: (gst_avi_demux_stream_init),
- (gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
- * gst/avi/gstavimux.c: (gst_avimux_stop_file):
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c: (gst_goom_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/intfloat/gstfloat2int.c: (gst_float2int_loop):
- * gst/intfloat/gstint2float.c: (gst_int2float_chain_gint16):
- * gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
- (gst_ebml_read_element_length), (gst_ebml_read_element_data),
- (gst_ebml_read_seek), (gst_ebml_read_uint), (gst_ebml_read_sint),
- (gst_ebml_read_float), (gst_ebml_read_header):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
- (gst_matroska_demux_parse_blockgroup):
- * gst/modplug/gstmodplug.cc:
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- * gst/oneton/gstoneton.c: (gst_oneton_chain):
- * gst/silence/gstsilence.c: (gst_silence_get):
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/smpte/gstsmpte.c: (gst_smpte_loop):
- * gst/speed/gstspeed.c: (speed_loop):
- * gst/tags/gstvorbistag.c: (gst_vorbis_tag_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * gst/volenv/gstvolenv.c: (gst_volenv_chain):
- * gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
- (gst_wavparse_loop):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_open),
- (dxr3audiosink_set_mode_pcm), (dxr3audiosink_set_mode_ac3),
- (dxr3audiosink_close):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_open), (dxr3spusink_close):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_open),
- (dxr3videosink_close), (dxr3videosink_write_data):
- * sys/oss/gstosselement.c: (gst_osselement_open_audio):
- * sys/oss/gstosselement.h:
- * sys/oss/gstosssink.c: (gst_osssink_get_type), (gst_osssink_init),
- (gst_osssink_chain):
- * sys/oss/gstosssrc.c: (gst_osssrc_get):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_buffer_free):
- * sys/v4l/v4l-overlay_calls.c: (gst_v4l_set_window),
- (gst_v4l_enable_overlay):
- * sys/v4l/v4l_calls.c: (gst_v4l_get_capabilities), (gst_v4l_open),
- (gst_v4l_set_chan_norm), (gst_v4l_get_signal),
- (gst_v4l_get_frequency), (gst_v4l_set_frequency),
- (gst_v4l_get_picture), (gst_v4l_set_picture), (gst_v4l_get_audio),
- (gst_v4l_set_audio):
- * sys/v4l/v4l_calls.h:
- * sys/v4l/v4lmjpegsink_calls.c: (gst_v4lmjpegsink_sync_thread),
- (gst_v4lmjpegsink_queue_frame), (gst_v4lmjpegsink_set_playback),
- (gst_v4lmjpegsink_playback_init),
- (gst_v4lmjpegsink_playback_start):
- * sys/v4l/v4lmjpegsrc_calls.c: (gst_v4lmjpegsrc_queue_frame),
- (gst_v4lmjpegsrc_sync_next_frame), (gst_v4lmjpegsrc_set_capture),
- (gst_v4lmjpegsrc_set_capture_m), (gst_v4lmjpegsrc_capture_init),
- (gst_v4lmjpegsrc_requeue_frame):
- * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_queue_frame),
- (gst_v4lsrc_sync_frame), (gst_v4lsrc_capture_init),
- (gst_v4lsrc_requeue_frame), (gst_v4lsrc_try_palette):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get):
- * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
- (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
- (gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
- (gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
- (gst_v4l2src_capture_stop):
- * sys/vcd/vcdsrc.c: (vcdsrc_open_file):
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_buffer_new),
- (gst_xvideosink_sinkconnect), (gst_xvideosink_chain),
- (gst_xvideosink_xwindow_new):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain):
- use new error signal, function and categories
-
-2004-01-18 Jeremy Simon <jesimon@libertysurf.fr>
+2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- * ext/Makefile.am:
- * ext/musicbrainz/gsttrm.c:
- * ext/musicbrainz/gsttrm.h:
- * ext/musicbrainz/Makefile.am:
- Add a trm plugin
+ gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
+ Original commit message from CVS:
+ * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
+ Support float audio in audiorate.
+ Use width rather than depth for selecting sample width.
-2004-01-18 Julien MOUTTE <julien@moutte.net>
+2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
- * sys/ximage/ximagesink.c: (gst_ximagesink_set_property),
- (gst_ximagesink_get_property), (gst_ximagesink_class_init): Adding
- synchronous property for debugging.
- * sys/ximage/ximagesink.h: Adding the synchronous boolean flag.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_set_property): Moving a pointer declaration to a
- smaller block, fixing indent.
+ gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
+ Original commit message from CVS:
+ * gst/videotestsrc/videotestsrc.h:
+ Use GLib types here (that way we don't have to include the
+ generated _stdint.h header, which makes life easier for win32
+ folks that don't use autotools for the build) (#325990, patch
+ by: Sergey Scobich).
-2004-01-16 David Schleef <ds@schleef.org>
+2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/videofilter/gstvideobalance.c: Fix regression; changing a
- property affects the video stream.
- * sys/xvimage/xvimagesink.c:
- * sys/xvimage/xvimagesink.h:
- Add synchronous property for debugging. Should probably be
- disabled in non-CVS builds. Make sure that the Xv attribute
- exists before we set it (crash!). Fix a silly float bug that
- caused colorbalance to just not work.
+ gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
+ (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (wait_segment):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Name (private) union, makes Forte compiler happy (this time
+ for real) (#324900).
-2004-01-17 Christian Schaller <Uraeus@gnome.org>
+2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
- * tools/gst-launch-ext.in - update for new plugins
+ gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ Link against libgstinterfaces, needed for mixer
+ and property probe stuff.
-2004-01-16 David Schleef <ds@schleef.org>
+2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect): Fix use of
- already-freed caps.
+ gst-libs/gst/Makefile.am:
+ Original commit message from CVS:
+ * gst-libs/gst/Makefile.am:
-2994-01-16 Christian Schaller <Uraeus@gnome.org>
+2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
- * Update spec for new colorspace plugin and libcaca plugin
- * Fix compilation of libcaca plugin (clock -> id)
+ gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
+ Original commit message from CVS:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/audio/mixerutils.c:
+ (gst_audio_mixer_filter_do_filter),
+ (gst_audio_mixer_filter_check_element),
+ (gst_audio_mixer_filter_probe_feature),
+ (element_factory_rank_compare_func),
+ (gst_audio_default_registry_mixer_filter):
+ * gst-libs/gst/audio/mixerutils.h:
+ Add gst_audio_default_registry_mixer_filter() utility
+ function.
-2004-01-16 Julien MOUTTE <julien@moutte.net>
+2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_update_colorbalance),
- (gst_xvimagesink_xcontext_get), (gst_xvimagesink_change_state),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_init), (gst_xvimagesink_class_init): Implementing
- correct colorbalance properties. They can now be set when the element
- is still in NULL state. The values will be committed to the Xv Port
- when xcontext is initialized.
- * sys/xvimage/xvimagesink.h: Added hue, saturation, contrast,
- brightness int values in the GstXvImagesink structure.
+ gst/audioresample/resample.h: As before, but for o_buf
+ Original commit message from CVS:
+ * gst/audioresample/resample.h:
+ As before, but for o_buf
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
- * gst-libs/gst/Makefile.am:
- restructure so having local patches works easier.
+ gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
+ Original commit message from CVS:
+ * gst/audioresample/resample.h:
+ Declare struct _ResampleState.buffer as unsigned char *, not void *,
+ since we do arithmetic on it.
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
- * ext/mpeg2enc/Makefile.am:
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- Bugfix with respect to EOS handling.
+ gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
+ (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
+ (gst_ring_buffer_pause), (wait_segment):
+ * gst-libs/gst/audio/gstringbuffer.h:
+ Sun's Forte compiler doesn't seem to like anonymous structs,
+ so use same setup as in GstBaseSrc (fixes #324900).
-2004-01-16 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect):
- Link with right caps (else, it segfaults).
- * ext/mplex/gstmplexjob.cc:
- Fix for slight API change in 1.6.1.93 release of mjpegtools.
+ move old example to tests/examples/volume/volune.c
+ Original commit message from CVS:
+ * configure.ac:
+ * gst/volume/Makefile.am:
+ * gst/volume/demo.c:
+ move old example to tests/examples/volume/volune.c
+ * tests/examples/Makefile.am:
+ * tests/examples/seek/seek.c: (main):
+ change window-close event from "delete-event" to "destroy"
+ * tests/examples/volume/Makefile.am:
+ * tests/examples/volume/volume.c: (value_changed_callback),
+ (setup_gui), (message_received), (eos_message_received), (main):
+ fix event handling and bus usage
-2004-01-15 David Schleef <ds@schleef.org>
+2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
+ (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
+ (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
+ (gst_audio_test_src_create_square),
+ (gst_audio_test_src_create_saw),
+ (gst_audio_test_src_create_triangle),
+ (gst_audio_test_src_create_silence),
+ (gst_audio_test_src_create_white_noise),
+ (gst_audio_test_src_create_pink_noise),
+ (gst_audio_test_src_init_sine_table),
+ (gst_audio_test_src_create_sine_table),
+ (gst_audio_test_src_change_wave),
+ (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
+ (gst_audio_test_src_create), (gst_audio_test_src_set_property):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ update to basesrc changes, implement segmented seeking and eos handling,
+ add a 'sine-tab' waveform for performance critical playback
+
+2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ po/POTFILES.in: ... and this time the other modified file that I missed last time.
+ Original commit message from CVS:
+ * po/POTFILES.in:
+ ... and this time the other modified file that I missed last time.
- * gst-libs/gst/audio/Makefile.am:
- Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
- from the template.
- * gst-libs/gst/audio/gstaudiofilter.c:
- * gst-libs/gst/audio/gstaudiofilter.h:
- Add bytes_per_sample and size and n_samples calculation.
- * gst-libs/gst/audio/gstaudiofilterexample.c:
- Remove, now autogenerated.
- * gst-libs/gst/audio/gstaudiofiltertemplate.c:
- Moved from gstaudiofilterexample, object name changed, code added
- so that it actually works.
- * gst-libs/gst/audio/make_filter:
- Script to build an audiofilter subclass from the template.
- * gst/colorspace/Makefile.am:
- * gst/colorspace/yuv2yuv.c:
- Remove file, since it's GPL, and we don't use it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): Making both of
- them use the buffer free function to test how the buffer was allocated.
-
-2004-01-15 David Schleef <ds@schleef.org>
-
- * ext/esd/esdsink.c: (gst_esdsink_class_init): Remove property
- that handles osssink fallback.
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
- (gst_audio_convert_getcaps):
- * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
- Add audio/x-qdm2 for QDM2 audio.
- * gst/sine/gstsinesrc.c: (gst_sinesrc_get):
- * gst/sine/gstsinesrc.h: Add example of how to implement tags.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps):
- Decrease minimum size to 16x16.
- * gst/wavparse/gstwavparse.c:
- Convert disabled pad template caps to new caps.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_chain): Throw element error when display cannot
- be opened. Increase minimum framerate to 1.0. Check the data
- free function on a buffer to make sure it is the type we expect
- before manipulating it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_init),
- (gst_videobalance_colorbalance_set_value): Implement passthru if
- settings are in the middle.
- * tools/gst-launch-ext.in: Stop using xvideosink, use ximagesink.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/videofilter/Makefile.am:
- * gst/volume/Makefile.am:
- Since we use videofilter symbols, link to it.
-
-2004-01-15 Julien MOUTTE <julien@moutte.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init): Setting
- mixer interface type to HARDWARE.
- * gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init): Adding a default
- type to SOFTWARE.
- * gst-libs/gst/mixer/mixer.h: Adding mixer interface type and macro.
- * gst-libs/gst/mixer/mixertrack.h: Adding mixertrack flag SOFTWARE.
- * gst/volume/gstvolume.c: (gst_volume_interface_supported),
- (gst_volume_interface_init), (gst_volume_list_tracks),
- (gst_volume_set_volume), (gst_volume_get_volume),
- (gst_volume_set_mute), (gst_volume_mixer_init),
- (gst_volume_dispose), (gst_volume_get_type), (volume_class_init),
- (volume_init): Implementing mixer interface.
- * gst/volume/gstvolume.h: Adding tracklist for mixer interface.
- * sys/oss/gstosselement.c: (gst_osselement_get_type),
- (gst_osselement_change_state): Removing some trailing commas in
- structures.
- * sys/oss/gstossmixer.c: (gst_ossmixer_interface_init): Setting mixer
- interface type to HARDWARE.
- * sys/v4l/gstv4lcolorbalance.c:
- (gst_v4l_color_balance_interface_init): Setting colorbalance interface
- type to HARDWARE.
- * sys/v4l2/gstv4l2colorbalance.c:
- (gst_v4l2_color_balance_interface_init): Setting colorbalance
- interface type to HARDWARE.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use exactly the
- same code than ximagesink for event handling.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/snapshot/Makefile.am:
- * ext/snapshot/gstsnapshot.c: (gst_snapshot_sinkconnect),
- (gst_snapshot_chain):
- * ext/snapshot/gstsnapshot.h:
- This has to be a joke... Snapshot should be connected to a tee,
- colorspace element before it and EOS after that, where the other
- src of the tee receives normal data.
- The current way is *wrong*.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/hermes/gsthermescolorspace.c:
- Fix another compile error. Same as below.
-
-2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/colorspace/gstcolorspace.c:
- * gst/colorspace/yuv2yuv.c: (gst_colorspace_yuy2_to_i420),
- (gst_colorspace_i420_to_yv12):
- Fix compiling... Didn't test if it actually works.
-
-2004-01-15 David Schleef <ds@schleef.org>
+2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
- * configure.ac:
- * gst/colorspace/Makefile.am:
- * gst/colorspace/gstcolorspace.c:
- * gst/colorspace/gstcolorspace.h:
- * gst/colorspace/yuv2rgb.c:
- * gst/colorspace/yuv2rgb.h:
- Duplicate the ext/hermes colorspace plugin, and remove Hermes
- code and GPL code. Fix for new caps negotiation. Rewrite
- much of the format handling code, and some of the conversion
- code. Basically, rewrote almost everything. This element
- handles I420, YV12 to RGB conversions.
- * ext/hermes/Makefile.am:
- * ext/hermes/gsthermescolorspace.c:
- Rename colorspace to hermescolorspace. Fix negotiation issues.
- Remove non-Hermes related code. This element handles lots of
- RGB to RGB conversions, but no YUV.
- * ext/hermes/gstcolorspace.c:
- * ext/hermes/gstcolorspace.h:
- * ext/hermes/rgb2yuv.c:
- * ext/hermes/yuv2rgb.c:
- * ext/hermes/yuv2rgb.h:
- * ext/hermes/yuv2rgb_mmx16.s:
- * ext/hermes/yuv2yuv.c:
- * ext/hermes/yuv2yuv.h:
- Remove old code.
-
-2004-01-14 Colin Walters <walters@verbum.org>
-
- * ext/mad/gstid3tag.c (gst_id3_tag_chain): Don't nego caps if
- they've already been.
-
-2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_id3_tag_do_caps_nego):
- assume tag mode when pad is not connected
-
-2004-01-15 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- Don't update the time of the clock
- (gst_alsa_sink_loop):
- sync to the clock given to alsasink, not the own clock
- * sys/oss/gstosssink.c: (gst_osssink_chain):
- sync to the clock
- (gst_osssink_change_state):
- activate the clock
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
- remove bogus code that made DISCONT events unhandled
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_video_caps):
- explicitly case to double in _set_simple. (fixes 2nd warning in bug
- #131502)
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_read_object_header),
- (gst_asf_demux_handle_sink_event), (gst_asf_demux_audio_caps),
- (gst_asf_demux_add_audio_stream), (gst_asf_demux_video_caps):
- convert g_warning because of wrong asf data to GST_WARNINGs (fixes
- 2nd warning in bug #131502)
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_init),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value): Fixing videobalance ranges
- for colorbalance interface implementation.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
- (gst_ximagesink_set_property), (gst_ximagesink_get_property),
- (gst_ximagesink_dispose), (gst_ximagesink_init),
- (gst_ximagesink_class_init): Adding DISPLAY property.
- * sys/ximage/ximagesink.h: Adding display_name to store display.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
- (gst_xvimagesink_dispose), (gst_xvimagesink_init),
- (gst_xvimagesink_class_init): Adding DISPLAY property and colorbalance
- properties (they still need polishing though for gst-launch use : no
- xcontext yet, i ll do that tomorrow).
- * sys/xvimage/xvimagesink.h: Adding display_name to store display.
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
- (gst_play_set_location), (gst_play_set_visualization): Preparing
- switch integration, adding videobalance in the pipeline.
-
-2004-01-14 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalance.c:
- (gst_color_balance_class_init): Adding a default type.
- * gst-libs/gst/colorbalance/colorbalance.h: Adding a macro to access
- the type.
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_get_type),
- (gst_videobalance_dispose), (gst_videobalance_class_init),
- (gst_videobalance_init), (gst_videobalance_interface_supported),
- (gst_videobalance_interface_init),
- (gst_videobalance_colorbalance_list_channels),
- (gst_videobalance_colorbalance_set_value),
- (gst_videobalance_colorbalance_get_value),
- (gst_videobalance_colorbalance_init): Implementing colorbalance
- interface.
- * gst/videofilter/gstvideobalance.h: Adding colorbalance channels
- list.
- * sys/ximage/ximagesink.c: (gst_ximagesink_set_xwindow_id): Fixing a
- bug which was triggering a BadAccess X error when setting an overlay
- before pad was really negotiated.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_colorbalance_init):
- Using the colorbalance type macro.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/flac/gstflacenc.c: (gst_flacenc_set_metadata),
- (gst_flacenc_chain):
- handle tags correctly
- * gst/tags/gstid3tag.c: (gst_tag_list_new_from_id3v1):
- extract ID3v1 tags correctly
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/typefind/gsttypefindfunctions.c: (matroska_type_find),
- (plugin_init):
- Improve matroska typefinding for odd-typed headers...
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
- Fix for using incremental number on padnames.
-
-2004-01-14 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- Set category to divx4linux instead of divx (too generic).
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt), (gst_wavparse_handle_sink_event),
- (gst_wavparse_loop), (gst_wavparse_change_state):
- * gst/wavparse/gstwavparse.h:
- fix parsing of WAV files with non-standard fmt-tag size and fix
- skipping of unrecognized chunks... Someone please fix this thing
- to use rifflib so all this is automated.
- * sys/v4l/Makefile.am:
- * sys/v4l2/Makefile.am:
- Add X_CFLAGS because we depend on X (for overlay).
-
-2004-01-14 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/mpeg2dec/gstmpeg2dec.c:
- Don't issue a timestamp unless we tagged the frame
- with a PTS.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_tick_callback):
- Query the audio element to get the time, not the clock. We're
- interested in the element's time here.
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/aalib/gstaasink.c: (gst_aasink_chain):
- * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
- * ext/esd/esdsink.c: (gst_esdsink_chain):
- * ext/libcaca/gstcacasink.c: (gst_cacasink_chain):
- * ext/mas/massink.c: (gst_massink_chain):
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_chain):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_index),
- (gst_matroska_demux_parse_metadata):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop),
- (gst_mpeg_parse_release_locks):
- * gst/tcp/gsttcpsink.c: (gst_tcpsink_chain):
- * gst/udp/gstudpsink.c: (gst_udpsink_chain):
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
- (gst_osssink_change_state):
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvideo/xvideosink.c: (gst_xvideosink_chain),
- (gst_xvideosink_release_locks):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
- use element time.
- * ext/alsa/gstalsaclock.c: (gst_alsa_clock_start),
- (gst_alsa_clock_stop):
- * gst-libs/gst/audio/audioclock.c: (gst_audio_clock_set_active),
- (gst_audio_clock_get_internal_time):
- simplify for use with new clocking code.
- * testsuite/alsa/Makefile.am:
- * testsuite/alsa/sinesrc.c: (sinesrc_init), (sinesrc_force_caps):
- fix testsuite for new caps system
-
-2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/flac/gstflacdec.c: (gst_flacdec_update_metadata):
- * ext/flac/gstflacenc.c: (add_one_tag):
- length is already host endian, no need to convert. Fixes playback of
- tagged files on PPC. (bug #128384)
-
-2004-01-13 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalance.h: Adding a type to the
- colorbalance interface stating if it is hardware based or software
- based.
- * gst/videofilter/gstvideobalance.c: (gst_videobalance_planar411):
- Removing a trailing comma.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_colorbalance_init): Integrating a patch from Jon
- Trowbridge <trow@ximian.com> querying Xv adaptor for min/max value as
- the documentation seems to be wrong on the -1000 to 1000 interval.
-
-2004-01-12 David Schleef <ds@schleef.org>
-
- * gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
- (gst_efence_buffer_alloc), (gst_fenced_buffer_new),
- (gst_fenced_buffer_default_free), (gst_fenced_buffer_default_copy):
- Fix negotiation. Add a bufferalloc function for the sink pad,
- and generally clean up some of the code.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/colorbalance/colorbalancechannel.c:
- (gst_color_balance_channel_dispose): Adding safety check in dispose
- method.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_xcontext_clear),
- (gst_xvimagesink_interface_supported),
- (gst_xvimagesink_colorbalance_list_channels),
- (gst_xvimagesink_colorbalance_set_value),
- (gst_xvimagesink_colorbalance_get_value),
- (gst_xvimagesink_colorbalance_init), (gst_xvimagesink_get_type):
- Adding colorbalance interface support to set XV parameters such as
- HUE, BRIGHTNESS, CONTRAST, SATURATION.
- * sys/xvimage/xvimagesink.h: Adding the channels list for colorbalance
- interface.
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_type),
- (gst_audio_convert_class_init), (gst_audioconvert_getcaps),
- (gst_audio_convert_init), (gst_audio_convert_set_property),
- (gst_audio_convert_get_property), (gst_audio_convert_chain),
- (gst_audio_convert_link),
- (gst_audio_convert_buffer_to_default_format),
- (gst_audio_convert_buffer_from_default_format), (plugin_init):
- - implement _getcaps and use it
- - improve linking
- - remove float caps since no float conversion is actually done
- - remove properties and arguments that were to be used for rate
- conversion
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-libs/gst/audio/audio.c: (_gst_audio_structure_set_list),
- (gst_audio_structure_set_int):
- * gst-libs/gst/audio/audio.h:
- add helper functions for _getcaps matching the standard audio
- templates
-
-2004-01-12 David Schleef <ds@schleef.org>
-
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
- Test that pad is negotiated before getting its caps.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * gst-libs/gst/play/gstplay.c: (gst_play_get_sink_element): When
- analyzing the pads of an element the bin is mostly in READY state so
- no caps were negotiated. This helper function needs to work with
- _get_caps directly then. I was not freeing them though, added that to
- fix the mem leak.
-
-2004-01-12 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): Fixing the
- direct put buffers detection. I prefer checking GST_BUFFER_PRIVATE
- than the free_func.
-
-2004-01-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * sys/oss/gstossaudio.c: (plugin_init):
- * sys/oss/gstosselement.c: (gst_osselement_sync_parms):
- * sys/oss/gstosselement.h:
- make an oss debugging category
- make failure more descriptive
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpeg.c:
- * ext/ffmpeg/gstffmpegcodecmap.c:
- * ext/ffmpeg/gstffmpegdec.c:
- * ext/ffmpeg/gstffmpegenc.c:
- * ext/ffmpeg/gstffmpegprotocol.c:
- * ext/gdk_pixbuf/gstgdkanimation.c:
- * ext/jpeg/gstjpeg.c:
- * ext/libpng/gstpng.c:
- * ext/mpeg2dec/perftest.c:
- * ext/speex/gstspeex.c:
- * gst-libs/gst/resample/dtos.c:
- * gst/intfloat/gstintfloatconvert.c:
- * gst/oneton/gstoneton.c:
- * gst/rtjpeg/RTjpeg.c:
- * gst/rtp/gstrtp.c:
- * sys/dxr3/dxr3init.c:
- * sys/glsink/gstgl_nvimage.c:
- * sys/glsink/gstgl_pdrimage.c:
- * sys/glsink/gstglsink.c:
- * testsuite/gst-lint:
- Make sure everybody wraps #include "config.h" in #ifdef HAVE_CONFIG_H
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_set_caps):
- * ext/faac/gstfaac.c: (gst_faac_sinkconnect):
- * ext/gdk_pixbuf/gstgdkanimation.c:
- (gst_gdk_animation_iter_create_pixbuf):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_chain):
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- * ext/jack/gstjack.c: (gst_jack_change_state):
- * ext/xvid/gstxviddec.c: (gst_xviddec_sink_link):
- * gst-libs/gst/play/gstplay.c: (gst_play_get_sink_element):
- * gst-libs/gst/play/play.c: (gst_play_get_sink_element):
- * gst/videofilter/gstvideofilter.c:
- (gst_videofilter_set_output_size):
- Remove all usage of gst_pad_get_caps(), and replace it with
- gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
-
-2004-01-11 David Schleef <ds@schleef.org>
+ gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
+ Original commit message from CVS:
+ * gst/playback/gstdecodebin.c: (new_pad):
+ Fix non-C89 variable declaration not at the start of a block. Should
+ help some compilers.
- * configure.ac:
- * ext/Makefile.am: Fixes to make ext/libcaca compile.
- * ext/divx/gstdivxdec.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link),
- (gst_gdk_pixbuf_init), (gst_gdk_pixbuf_chain): Make gdkpixbufdec
- handle images that span multiple buffers. Now work with both
- filesrc ! gdkpixbufdec and qtdemux ! gdkpixbufdec.
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
- * ext/libcaca/gstcacasink.h: Fixes needed due to recent
- video/video.h changes
- * ext/xvid/gstxvid.c: (gst_xvid_csp_to_caps): same
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free): Use buffer free function instead
- of GstData free function.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_buffer_free):
- same.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/v4l2/gstv4l2element.c: (gst_v4l2element_class_init),
- (gst_v4l2element_dispose), (gst_v4l2element_set_property),
- (gst_v4l2element_get_property):
- * sys/v4l2/v4l2_calls.c: (gst_v4l2_set_defaults), (gst_v4l2_open):
- add norm, channel and frequency properties.
- * sys/v4l2/gstv4l2tuner.c:
- fixes for tuner interface changes
- * sys/v4l2/gstv4l2element.h:
- * sys/v4l2/gstv4l2src.c:
- * sys/v4l2/gstv4l2src.h:
- * sys/v4l2/v4l2src_calls.c:
- * sys/v4l2/v4l2src_calls.h:
- rework v4l2src to work with saa1734 cards and allow mmaped buffers.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst-libs/gst/tuner/tuner.c: (gst_tuner_class_init),
- (gst_tuner_find_norm_by_name), (gst_v4l2_find_channel_by_name),
- (gst_tuner_channel_changed), (gst_tuner_norm_changed),
- (gst_tuner_frequency_changed), (gst_tuner_signal_changed):
- * gst-libs/gst/tuner/tuner.h:
- GObjects aren't const.
- Add find_by_name functions.
- Add checks to _changed functions.
- * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_get_channel),
- (gst_v4l_tuner_get_norm):
- Fixes for above.
-
-2004-01-12 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst-libs/gst/video/video.h:
- Fix caps template names to be understandable.
- Prefix everything with GST_VIDEO.
- * ext/aalib/gstaasink.c:
- * ext/divx/gstdivxdec.c:
- * ext/divx/gstdivxenc.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.c:
- * ext/hermes/gstcolorspace.c: (gst_colorspace_base_init):
- * ext/jpeg/gstjpegdec.c: (raw_caps_factory):
- * ext/jpeg/gstjpegenc.c: (raw_caps_factory):
- * ext/libcaca/gstcacasink.c:
- * ext/libpng/gstpngenc.c: (raw_caps_factory):
- * ext/snapshot/gstsnapshot.c:
- * ext/swfdec/gstswfdec.c:
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- * gst/chart/gstchart.c:
- * gst/deinterlace/gstdeinterlace.c:
- * gst/effectv/gsteffectv.c:
- * gst/flx/gstflxdec.c: (gst_flxdec_loop):
- * gst/goom/gstgoom.c:
- * gst/median/gstmedian.c:
- * gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
- (gst_monoscope_srcconnect), (gst_monoscope_chain):
- * gst/overlay/gstoverlay.c:
- * gst/smooth/gstsmooth.c:
- * gst/smpte/gstsmpte.c:
- * gst/synaesthesia/gstsynaesthesia.c:
- * gst/videocrop/gstvideocrop.c:
- * gst/videodrop/gstvideodrop.c:
- * gst/y4m/gsty4mencode.c:
- * sys/qcam/gstqcamsrc.c:
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
- Make them work with new video.h file.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
- Make it work with new buffer allocation system.
-
-2004-01-11 Julien MOUTTE <julien@moutte.net>
-
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * ext/hermes/gstcolorspace.c: (gst_colorspace_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Fixing the
- pad_alloc_buffer implementation to use ->srcpad
- * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_free),
- (gst_ximagesink_buffer_alloc): Now only use GST_BUFFER_PRIVATE to keep
- a reference to everything we need.
- * sys/ximage/ximagesink.h: adding a reference to the sink in the image.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
- (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free),
- (gst_xvimagesink_buffer_alloc): Now only use GST_BUFFER_PRIVATE to keep
- a reference to everything we need.
- * sys/xvimage/xvimagesink.h: adding a reference to the sink in the image
-
-2004-01-11 David Schleef <ds@schleef.org>
-
- * ext/divx/gstdivxenc.c: remove bogus gst_caps_is_fixed() test
- * gst/debug/efence.c: (gst_efence_chain), (gst_fenced_buffer_new),
- (gst_fenced_buffer_default_copy): Fix for rename of buffer private
- structure members.
- * gst/effectv/gstwarp.c: (gst_warptv_setup): Don't reset the time
- value during a resize/renegotiation.
- * gst/videofilter/gstvideofilter.c: (gst_videofilter_chain): use
- gst_pad_alloc_buffer();
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free): Fix for rename of buffer private
- structure members.
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_buffer_free):
- Fix for rename of buffer private structure members.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
- Fix for rename of buffer private structure members.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
- Fix for rename of buffer private structure members.
-
-2004-01-11 Arwed v. Merkatz <v.merkatz@gmx.net>
-
- reviewed by: David Schleef <ds@schleef.org>
-
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstgamma.c: Gamma correction filter. Modified
- from the patch by ds to fit in with recent make_filter changes.
-
-2004-01-11 Julien MOUTTE <julien@moutte.net>
-
- * configure.ac: Adding examples/switch/Makefile
- * examples/Makefile.am: Adding examples/switch
- * examples/switch/Makefile.am: Adding switcher example.
- * examples/switch/switcher.c: (got_eos), (idle_iterate),
- (switch_timer), (main): Adding an example demonstrating switch usage
- with 2 videotestsrc showing different patterns.
- * gst/switch/gstswitch.c: (gst_switch_request_new_pad),
- (gst_switch_init): Fixing switch with the new caps system.
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ And now fix 'make distcheck' (builddir != srcdir)
- * gst-libs/gst/video/video.h:
- Fix 32bit caps. Issue remaining: The macro names are chosen poorly.
- They should probably be like
- GST_VIDEO_PAD_TEMPLATE_CAPS_{RGB,BGR,RGBx,BGRx}.
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (qtdemux_parse_trak):
- fix audio chunk size/timestamp calculation
-
-2004-01-11 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps):
- fix SVQ3 caps
-
-2004-01-11 Steve Baker <steve@stevebaker.org>
-
- * gst/effectv/gstaging.c: (gst_agingtv_get_type),
- (gst_agingtv_base_init), (gst_agingtv_class_init),
- (gst_agingtv_init), (gst_agingtv_setup), (gst_agingtv_rgb32),
- (gst_agingtv_set_property), (gst_agingtv_get_property):
- Port agingTV to videofilter
-
-2004-01-09 Julien MOUTTE <julien@moutte.net>
-
- * ext/hermes/gstcolorspace.c: (gst_colorspace_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
-
-2004-01-09 Julien MOUTTE <julien@moutte.net>
-
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * gst-libs/gst/xoverlay/xoverlay.c:
- (gst_x_overlay_got_desired_size): Updating doc for the xid being 0.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
- * sys/ximage/ximagesink.c: (gst_ximagesink_chain),
- (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_init): Implementing
- the bufferalloc_function to replace bufferpools, fixing the XOverlay
- interface implementation to handle xid being 0 and fix some bugs
- triggered by Benjamin's testcase.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
- (gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_init): Implementing
- the bufferalloc_function to replace bufferpools, fixing the XOverlay
- interface implementation to handle xid being 0 and fix some bugs
- triggered by Benjamin's testcase.
-
-2004-01-09 David Schleef <ds@schleef.org>
-
- * ext/librfb/gstrfbsrc.c: Hacking. Added actual decoding and
- mouse pointer events. It works.
-
-2004-01-09 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/divx/gstdivxenc.c: (gst_divxenc_init):
- Use explicit caps - fix capsnego.
- * ext/xvid/gstxviddec.c:
- * ext/xvid/gstxvidenc.c:
- Remove macro-inside-macro which caused compile errors.
- * gst-libs/gst/riff/riff-read.c: (gst_riff_read_header):
- Error out if it's not a RIFF file. Else we error out without
- gst_element_error() which is not good...
-
-2004-01-08 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect):
- Fix pad_link function to handle formats that ffmpeg returns
- as multiple caps structures.
- * gst/videofilter/gstvideofilter.c: (gst_videofilter_chain):
- Only complain if source buffer is _smaller_ than expected.
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
- (gst_videoscale_handle_src_event): Resize navigation events
- when passing them upstream.
- * gst/videotestsrc/gstvideotestsrc.c:
- * gst/videotestsrc/gstvideotestsrc.h:
- * gst/videotestsrc/videotestsrc.c:
- * gst/videotestsrc/videotestsrc.h:
- Rewrite many of the buffer painting functions to handle odd
- sizes (for many formats, size%4!=0 or size%8!=0). Most have
- been verified to work with my video card.
- * testsuite/gst-lint: Add check for elements calling
- gst_pad_get_caps() instead of gst_pad_get_allowed_caps().
-
-2004-01-08 David Schleef <ds@schleef.org>
-
- * gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
- (gst_videodrop_link), (gst_videodrop_init): Fix negotiation.
-
-2004-01-08 Julien MOUTTE <julien@moutte.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents): A
- configure event is not emiting the desired size signal. That fixes
- aspect ratio issues with gst-player.
-
-2004-01-08 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/median/gstmedian.c: (gst_median_link), (gst_median_init):
- Fix capsnego.
-
-2004-01-08 Julien MOUTTE <julien@moutte.net>
-
- * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create): Using XOverlay
- public method to fire size signal.
-
-2004-01-07 Julien MOUTTE <julien@moutte.net>
-
- * examples/gstplay/Makefile.am: Adding the interface library.
- * gst-libs/gst/play/Makefile.am: Adding the interface library.
- * gst-libs/gst/play/gstplay.c: (gst_play_set_video_sink): Connecting to the XOverlay size signal instead of GstVideoSink.
- * gst-libs/gst/play/gstplay.h: Including the XOverlay interface to check
- GST_IS_X_OVERLAY before signal connect.
- * gst-libs/gst/video/gstvideosink.c: (gst_videosink_class_init):
- Removing the have_video_size signal.
- * gst-libs/gst/video/gstvideosink.h: Removing the have_video_size signal
- and associated public method.
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_sinkconnect): Using XOverlay public method to fire size
- signal.
- * sys/xvideo/xvideosink.c: (gst_xvideosink_sinkconnect),
- (gst_xvideosink_xwindow_new): Using XOverlay public method to fire size
- signal.
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_sinkconnect):
- Using XOverlay public method to fire size signal.
-
-2004-01-07 David Schleef <ds@schleef.org>
-
- * gst/videofilter/Makefile.am:
- * gst/videofilter/gstvideotemplate.c:
- * gst/videofilter/make_filter:
- Create gstvideoexample.c in a srcdir!=builddir friendly way.
- Convert make_filter to /bin/sh script.
-
-2004-01-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst/modplug/gstmodplug.cc: fix element description
-
-2004-01-07 Julien MOUTTE <julien@moutte.net>
-
- * examples/gstplay/player.c: (got_time_tick), (got_stream_length),
- (got_video_size): Adding some new lines in g_print calls.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy), (gst_ximagesink_xwindow_resize),
- (gst_ximagesink_handle_xevents), (gst_ximagesink_fixate),
- (gst_ximagesink_sinkconnect), (gst_ximagesink_change_state),
- (gst_ximagesink_chain), (gst_ximagesink_buffer_new),
- (gst_ximagesink_set_xwindow_id), (gst_ximagesink_get_desired_size):
- Complete code review, reverting some stuff i disagree with, adding
- some fixes : time synchronization on invalid timestamps, renegotiation
- of private window.
- * sys/ximage/ximagesink.h:
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_destroy),
- (gst_xvimagesink_xwindow_resize), (gst_xvimagesink_handle_xevents),
- (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
- (gst_xvimagesink_fixate), (gst_xvimagesink_sinkconnect),
- (gst_xvimagesink_change_state), (gst_xvimagesink_chain),
- (gst_xvimagesink_buffer_new),
- (gst_xvimagesink_navigation_send_event),
- (gst_xvimagesink_set_xwindow_id),
- (gst_xvimagesink_get_desired_size),
- (gst_xvimagesink_xoverlay_init): Complete code review, reverting some
- stuff i disagree with, adding some fixes : Renegotiation of private
- window, implementing get_desired_size.
-
-2004-01-07 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/audiofile/gstafsink.c: (gst_afsink_init), (gst_afsink_chain),
- (gst_afsink_handle_event):
- * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
- * gst/avi/gstavimux.c: (gst_avimux_request_new_pad):
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_init):
- Fix for instantiate-test (see core). Also remove dead code from
- jpegenc (which still needs fixing, but that's lower on my TODO
- list...).
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_getcaps):
- Never return NULL as caps.
-
-2004-01-07 David Schleef <ds@schleef.org>
+2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
+ Original commit message from CVS:
+ * configure.ac:
+ * ext/cdparanoia/Makefile.am:
+ * ext/cdparanoia/gstcdparanoia.c:
+ * ext/cdparanoia/gstcdparanoia.h:
+ * ext/cdparanoia/gstcdparanoiasrc.c:
+ (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
+ (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
+ (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
+ (gst_cd_paranoia_paranoia_callback),
+ (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
+ (gst_cd_paranoia_src_set_property),
+ (gst_cd_paranoia_src_get_property), (plugin_init):
+ * ext/cdparanoia/gstcdparanoiasrc.h:
+ New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
+ plugin again (there are still fixes required to playbin to make
+ cdda:// uris work there).
- * configure.ac:
- * ext/Makefile.am:
- * ext/librfb/Makefile.am:
- * ext/librfb/gstrfbsrc.c:
- New source plugin based on librfb-0.1. RFB (remote framebuffer)
- is the protocol used by VNC.
-
-2004-01-07 David Schleef <ds@schleef.org>
-
- * gst/videofilter/gstvideotemplate.c:
- * gst/videofilter/gstvideotemplate.h:
- * gst/videofilter/make_filter:
- Merge videotemplate header into source file.
- * gst/effectv/Makefile.am:
- * gst/effectv/gsteffectv.c: (plugin_init):
- * gst/effectv/gstwarp.c:
- Make warpTV a subclass of videofilter.
-
-2004-01-07 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list):
- Add guard against invalid utf-8 conversions in mad. Just in case.
-
-2004-01-07 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * sys/oss/gstosssink.c: (gst_osssink_sink_fixate):
- Fix for bug shown by poisoning
-
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
- (gst_v4lmjpegsrc_buffer_free):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps),
- (gst_v4lsrc_srcconnect), (gst_v4lsrc_getcaps), (gst_v4lsrc_get),
- (gst_v4lsrc_buffer_free):
- Fix for removed bufferpools.
-
-2004-01-07 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/dv/gstdvdec.c: (gst_dvdec_loop):
- Fix caps negotiation.
-
- * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init),
- (dvdnavsrc_update_buttoninfo), (dvdnavsrc_get),
- (dvdnavsrc_get_event_mask), (dvdnav_handle_navigation_event),
- (dvdnavsrc_event):
- * ext/mpeg2dec/gstmpeg2dec.c:
- * gst-libs/gst/navigation/navigation.c:
- (gst_navigation_send_key_event), (gst_navigation_send_mouse_event):
- * gst-libs/gst/navigation/navigation.h:
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_handle_src_event):
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
- Super-simple first version of mouse and keyboard events. Clicking
- on a DVD menu now works, although it may not take you where you
- expected.
-
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate):
- These fixate functions were broken - they never actually
- fixated :)
+2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ tests/check/Makefile.am: Fix test case compilation.
+ Original commit message from CVS:
+ * tests/check/Makefile.am:
+ Fix test case compilation.
- * ext/shout/gstshout.c: (gst_icecastsend_base_init),
- (gst_icecastsend_init):
- fix for new caps system.
- * gst-libs/gst/mixer/mixertrack.h:
- * sys/oss/gstossmixer.c: (gst_ossmixer_build_list):
- Add 'master track' flag (for tools like ACME that only want to
- change the main volume).
+2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-07 Jan Schmidt <thaytan@mad.scientist.com>
+ gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
+ Original commit message from CVS:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_update_duration),
+ (gst_cdda_base_src_calculate_cddb_id):
+ An integer is not a string. Fix access to uninitialised variable.
+ * tests/check/Makefile.am:
+ Add cddabasesrc unit test; also actually enable the vorbis test.
+ * tests/check/generic/states.c:
+ Blacklist new cd audio elements as well.
+ * tests/check/libs/cddabasesrc.c:
+ Unit test for GstCddaBaseSrc (discid calculation mostly).
- * ext/xvid/gstxvid.c: (gst_xvid_structure_to_csp),
- (gst_xvid_csp_to_caps):
- * ext/xvid/gstxviddec.c: (gst_xviddec_src_getcaps):
- * ext/xvid/gstxvidenc.c:
- ifdef out ARGB type when it isn't available
- in xvidcore 1.0.0beta2
+2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
+ Original commit message from CVS:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * docs/libs/gst-plugins-base-libs.types:
+ Add docs for libgstcdda/GstCddaBaseSrc.
+ * gst-libs/gst/interfaces/mixertrack.h:
+ Do one struct member per line with a semicolon at the end, that way
+ even gtk-doc might parse it without complaining.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop):
- When we have received a new SCR right in the first buffer after
- a seek (so in the same cycle that handles the discont), we should
- handle the buffer instead of unreffing it, else we lose data.
+2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
-2004-01-06 Iain <iain@prettypeople.org>
+ Add new libgstcdda with GstCddaBaseSrc class.
+ Original commit message from CVS:
+ * configure.ac:
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/cdda/base64.c:
+ * gst-libs/gst/cdda/base64.h:
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
+ (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
+ (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
+ (gst_cdda_base_src_get_property),
+ (gst_cdda_base_src_get_track_from_sector),
+ (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
+ (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
+ (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
+ (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
+ (gst_cdda_base_src_uri_get_protocols),
+ (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
+ (gst_cdda_base_src_uri_handler_init),
+ (gst_cdda_base_src_setup_interfaces),
+ (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
+ (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
+ (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
+ (gst_cdda_base_src_add_tags),
+ (gst_cdda_base_src_add_index_associations),
+ (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
+ (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
+ (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
+ (gst_cdda_base_src_create):
+ * gst-libs/gst/cdda/gstcddabasesrc.h:
+ * gst-libs/gst/cdda/sha1.c:
+ * gst-libs/gst/cdda/sha1.h:
+ Add new libgstcdda with GstCddaBaseSrc class.
- * gst/intfloat/gstint2float.c (gst_int2float_link): Set the
- buffer-frames caps too.
+2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/oneton/gstoneton.c (gst_oneton_sink_connect): Only create the new
- caps that we need, don't destroy them all and rebuild them. And when
- creating src pads, use the src pad template rather than the sink...
+ ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
+ Original commit message from CVS:
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Use GstBaseSinkClass as parent_class member for class struct, not
+ GstBaseSink.
-2004-01-05 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead):
- Add pad to element *after* setting functions such as event handler.
- Without this, the scheduler (opt) will link pads, set the event
- handler from the default event function (dispatcher in gstpad.c)
- and *after* that, we will set our own event function, which will
- thus never be used (and thus mpegdemux doesn't handle events).
+ gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_start):
+ Add start method to reset running time and number of frames sent
+ when starting up (fixes #324696; patch by: Michal Benes).
+
+2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
+ Original commit message from CVS:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-base-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-base-plugins-sections.txt:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ Add docs stuff for gnomevfssrc and gnomevfssink.
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ Fix example pipeline in gtk-doc blurb.
-2004-01-04 David Schleef <ds@schleef.org>
+2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
+ Original commit message from CVS:
+ * ext/gnomevfs/Makefile.am:
+ * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
+ (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
+ (gst_gnome_vfs_handle_get_type), (plugin_init):
+ * ext/gnomevfs/gstgnomevfs.h:
+ * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
+ (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
+ (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
+ (gst_gnome_vfs_sink_set_property),
+ (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
+ (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
+ (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
+ (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
+ (gst_gnome_vfs_sink_uri_get_type),
+ (gst_gnome_vfs_sink_uri_get_protocols),
+ (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
+ (gst_gnome_vfs_sink_uri_handler_init):
+ * ext/gnomevfs/gstgnomevfssink.h:
+ Port gnomevfssink; add gtk-doc blurb.
+ * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
+ (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
+ (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
+ (gst_gnome_vfs_src_uri_get_type),
+ (gst_gnome_vfs_src_uri_get_protocols),
+ (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
+ (gst_gnome_vfs_src_uri_handler_init),
+ (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
+ (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
+ (gst_gnome_vfs_src_send_additional_headers_callback),
+ (gst_gnome_vfs_src_received_headers_callback),
+ (gst_gnome_vfs_src_push_callbacks),
+ (gst_gnome_vfs_src_pop_callbacks),
+ (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
+ (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
+ (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
+ * ext/gnomevfs/gstgnomevfssrc.h:
+ s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
+ file; add gtk-doc blurb with example pipelines.
- Fix the fixate functions to handle new prototype:
- * gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
- * gst/videotestsrc/gstvideotestsrc.c:
- (gst_videotestsrc_src_fixate):
- * sys/oss/gstosssink.c: (gst_osssink_sink_fixate):
- * sys/ximage/ximagesink.c: (gst_ximagesink_fixate):
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
+2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * sys/ximage/ximagesink.h:
- * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
- (gst_ximagesink_xwindow_destroy), (gst_ximagesink_sinkconnect),
- (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
- (gst_ximagesink_xoverlay_init):
- assorted fixes to make (re)embedding work
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect),
- (gst_ximagesink_get_desired_size):
- implement desired size additions to XOverlay
+=== release 0.10.1 ===
-2004-01-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
+2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_base_init),
- (gst_x_overlay_got_xwindow_id), (gst_x_overlay_get_desired_size),
- (gst_x_overlay_got_desired_size):
- * gst-libs/gst/xoverlay/xoverlay.h:
- Add optional "desired size" signal and querying.
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.1
+ Original commit message from CVS:
+ releasing 0.10.1
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix EBML-laced block parsing. Diffs are relative to previous
- lace, not the first lace. Thanks to Mosu from the Matroska
- team for detecting this.
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt), (gst_wavparse_getcaps),
- (gst_wavparse_handle_sink_event), (gst_wavparse_loop),
- (gst_wavparse_change_state):
- * gst/wavparse/gstwavparse.h:
- Quickfix for capsnego.
+ * ChangeLog:
+ * gst/typefind/gsttypefindfunctions.c:
+ iLBC30 and iLBC20 added to typefind.
+ Original commit message from CVS:
+ iLBC30 and iLBC20 added to typefind.
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * gst/wavenc/gstwavenc.c: (set_property), (gst_wavenc_init):
- Fix indenting, fix pad creation.
+ * ChangeLog:
+ * configure.ac:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * po/af.po:
+ * po/az.po:
+ * po/cs.po:
+ * po/en_GB.po:
+ * po/hu.po:
+ * po/it.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ prereleasing
+ Original commit message from CVS:
+ prereleasing
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/xvid/gstxviddec.c: (gst_xviddec_init),
- (gst_xviddec_src_getcaps), (gst_xviddec_src_link),
- (gst_xviddec_sink_link):
- Implement src_getcaps() so proper size caps is negotiated.
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ stop making fun of older compilers
+ Original commit message from CVS:
+ stop making fun of older compilers
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
- * ext/flac/gstflacdec.c: (gst_flacdec_loop):
- Finish flac decoder on EOS. See #116178.
+ gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
+ Original commit message from CVS:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ (gst_base_audio_sink_class_init):
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ (gst_base_audio_src_class_init):
+ update strings, values are in microseconds
+ change the default sink buffer time to something that is smaller
+ (to help software volume mixing have a slightly lower delay) but
+ still be acceptable on Wim's laptop
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
+ Original commit message from CVS:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
+ Made a quack, forgot to add DUCK to the riff video template.
+
+2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
+
+ ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
+ (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
+ (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
+ (gst_ogm_parse_chain):
+ Make sure pads are initialized correctly.
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
+ (gst_riff_create_video_template_caps):
+ Add a whole bunch of FOURCC <=> MimeType.
+ Extend the riff video pad template to support the newly added fourcc.
+
+2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
+
+ ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
+ (gst_ogg_demux_activate_chain):
+ Extra debug output when activating/deactivating chains.
+ * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
+ (is_demuxer_element), (try_to_link_1), (remove_element_chain),
+ (unlinked):
+ Remove a queue from our list when it becomes unlinked.
+ Don't add queues to elements in class 'Demux' if they
+ can only produce one pad
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_src_getcaps),
- (gst_matroska_demux_add_stream):
- * gst/matroska/matroska-ids.h:
- Add getcaps() function to fix capsnego...
+2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
-2004-01-04 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/video/gstvideosink.c: Add a debug category.
+ Original commit message from CVS:
+ 2005-12-18 Julien MOUTTE <julien@moutte.net>
+ * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
+ (gst_video_sink_get_type): Add a debug category.
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (gst_mpeg_demux_parse_packet), (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_parse_packhead):
- Fix more integer overflows. Again, see #126967.
+2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
+ Original commit message from CVS:
+ 2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
+ Handle downstream newsegment by sending our own newsegment before the
+ next buffer to be released. (#323900)
- * ext/mpeg2dec/gstmpeg2dec.c:
- Add support for mpeg2dec-0.4.0 (released two weeks ago). See
- #130416.
+2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
+ Original commit message from CVS:
+ 2005-12-17 Philippe Khalaf <burger@speedy.org>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ add queue delay to new segment as well (as opposed to just the first
+ buffer). (bug #322347)
- * configure.ac:
- * ext/xvid/gstxvid.c: (gst_xvid_init), (gst_xvid_error),
- (gst_xvid_structure_to_csp), (gst_xvid_csp_to_caps):
- * ext/xvid/gstxvid.h:
- * ext/xvid/gstxviddec.c: (gst_xviddec_class_init),
- (gst_xviddec_init), (gst_xviddec_setup), (gst_xviddec_chain),
- (gst_xviddec_src_link), (gst_xviddec_sink_link),
- (gst_xviddec_change_state):
- * ext/xvid/gstxviddec.h:
- * ext/xvid/gstxvidenc.c: (gst_xvidenc_profile_get_type),
- (gst_xvidenc_base_init), (gst_xvidenc_class_init),
- (gst_xvidenc_init), (gst_xvidenc_setup), (gst_xvidenc_chain),
- (gst_xvidenc_link), (gst_xvidenc_set_property),
- (gst_xvidenc_get_property), (gst_xvidenc_change_state):
- * ext/xvid/gstxvidenc.h:
- Update xvid plugin to latest xvid (1.0.0-beta3) API.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/rtp/rtp-packet.c:
- Add sys/types.h include, since OS X doesn't define in_addr_t
- in netinet/in.h, like it does on Linux (see #129600).
-
-2004-01-03 Thomas Canty <tommydal@optushome.com.au>
-
- reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_dispose):
- Correct logic of dispose function (see #129306).
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_pes):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
- (gst_mpeg_parse_init):
- * gst/mpegstream/gstmpegparse.h:
- Remove clock (which was never provided, i.e. dead code), and
- also fix integer overflows at high PTS values (see #126967).
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/flac/gstflacdec.c:
- * ext/libpng/gstpngenc.h:
- * ext/mikmod/gstmikmod.h:
- OS X fixes (see #126628).
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * ext/alsa/gstalsasrc.c: (gst_alsa_src_pad_factory),
- (gst_alsa_src_base_init): Remove bogus "src" request pad.
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_base_init),
- (gst_mpeg_parse_class_init): Move pad template registration
- to class_init, since the derived class (mpegdemux) doesn't
- want them.
-
-2004-01-03 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
- * sys/ximage/Makefile.am:
- * sys/xvideo/Makefile.am:
- * sys/xvimage/Makefile.am:
- Move interface libs from LDFLAGS to LIBADD, fix relocation errors
- after installation (see #127664).
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_init),
- (gst_ffmpegenc_connect): Negotiation fixes.
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_negotiate_format):
- Remove inappropriate gst_caps_free().
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Reenable Ronald's internal resize code, since the core handles
- it correctly now.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_init):
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init), (gst_v4lsrc_init):
- Fix pad template stuff.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * gst/matroska/ebml-read.c: (gst_ebml_read_sint):
- * gst/matroska/ebml-write.c: (gst_ebml_write_sint):
- fix signed integer reading/writing.
-
-2004-01-02 Benjamin Otte <in7y118@public.uni-hamburg.de>
-
- * ext/alsa/README:
- Remove outdated document
-
-2004-01-03 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst/cutter/gstcutter.c: (gst_cutter_init):
- src pad was being created twice - oops.
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Comment out internal resize. It doesn't handle the resulting
- XEvent internally, does another try_set_caps() which leads to
- a really nice loop.
- Real fix will come when Julien and Dave are awake. ;).
-
-2004-01-02 Ronald Bultje <rbultje@ronald.bitfreak.net>
-
- * ext/mpeg2enc/gstmpeg2enc.cc:
- fix const/nonconst compile issue.
-
-2004-01-02 David Schleef <ds@schleef.org>
-
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate),
- (gst_xvimagesink_sinkconnect), (gst_xvimagesink_init):
- Add fixate function and a check for bad formats.
-
-2004-01-01 David Schleef <ds@schleef.org>
-
- Negotiation fixes:
- * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
- (gst_audiofilter_init):
- * gst/debug/efence.c: (gst_efence_init):
- * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_link),
- (gst_deinterlace_init):
- * gst/volume/gstvolume.c: (volume_connect):
-
-2004-01-01 David Schleef <ds@schleef.org>
-
- Convert elements to use gst_pad_use_explicit_caps() where
- appropriate:
- * ext/a52dec/gsta52dec.c: (gst_a52dec_init), (gst_a52dec_reneg):
- * ext/audiofile/gstafparse.c: (gst_afparse_init),
- (gst_afparse_open_file):
- * ext/audiofile/gstafsrc.c: (gst_afsrc_init),
- (gst_afsrc_open_file):
- * ext/esd/esdmon.c: (gst_esdmon_init), (gst_esdmon_get):
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_init),
- (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_loop):
- * ext/flac/gstflacdec.c: (gst_flacdec_init), (gst_flacdec_write):
- * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init),
- (gst_gdk_pixbuf_chain):
- * ext/jpeg/gstjpegdec.c: (gst_jpegdec_init), (gst_jpegdec_link),
- (gst_jpegdec_chain):
- * ext/mad/gstmad.c: (gst_mad_init), (gst_mad_chain):
- * ext/mikmod/gstmikmod.c: (gst_mikmod_init),
- (gst_mikmod_negotiate):
- * ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_init),
- (gst_mpeg2dec_negotiate_format):
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/ogg/gstoggdemux.c: (gst_ogg_pad_push):
- * ext/speex/gstspeexdec.c: (gst_speexdec_init),
- (gst_speexdec_sinkconnect):
- * ext/swfdec/gstswfdec.c: (gst_swfdec_loop), (gst_swfdec_init):
- * ext/vorbis/vorbisfile.c: (gst_vorbisfile_init),
- (gst_vorbisfile_new_link):
- * gst/ac3parse/gstac3parse.c: (gst_ac3parse_init),
- (gst_ac3parse_chain):
- * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_audio_stream),
- (gst_asf_demux_setup_pad):
- * gst/auparse/gstauparse.c: (gst_auparse_init),
- (gst_auparse_chain):
- * gst/id3/gstid3types.c: (gst_id3types_loop):
- * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
- * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
- (mp1videoparse_parse_seq):
- * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_init),
- (bpf_from_header):
- * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead),
- (gst_mpeg_demux_parse_pes), (gst_mpeg_demux_lpcm_set_caps):
- * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init),
- (gst_mpeg_parse_send_data):
- * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
- (gst_qtdemux_add_stream):
- * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream):
- * gst/wavparse/gstwavparse.c: (gst_wavparse_init),
- (gst_wavparse_parse_fmt):
-
-2004-01-01 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ ext/libvisual/visual.c: change some char* into char[]
+ Original commit message from CVS:
+ * ext/libvisual/visual.c: (make_valid_name):
+ change some char* into char[]
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
+ (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ prepare to handle EOS and SEGMENT_DONE
- * configure.ac:
- Fix configure check for mpeg2enc. We need 1.6.1.93 instead of
- 1.6.1.92, since the pkg-config file of 1.6.1.92 is borked and
- it therefore uses the wrong include paths. Too bad... Note
- that 1.6.1.93 is not release yet. ;).
- Also add a check for mplex, which is now using the lib'ified
- mplex from mjpegtools, too.
- * ext/ffmpeg/gstffmpegcodecmap.c:
- Add codec_tag for 3ivx/xvid. For xvid, this should fix playback
- issues. I don't think ffmpeg handles 3ivx correctly, so this
- probably won't work. But it won't hurt either.
- * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_connect),
- (gst_ffmpegdec_chain):
- * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect),
- (gst_ffmpegenc_chain_audio):
- Fix memleak in audio encoding. Close codec if open fails, this
- calls the cleanup routines so we can re-use the context.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Fix pad template names/types, fix memory issue with getcaps().
- * ext/mpeg2enc/gstmpeg2encoder.cc:
- * ext/mpeg2enc/gstmpeg2encoder.hh:
- Fix compile issue with new caps system (const thingy).
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.hh:
- We read a first frame right on initing, so that we have a caps
- when we init the output. This caps is cached in padprivate and
- read as first frame.
- * ext/mplex/Makefile.am:
- * ext/mplex/gstmplex.cc:
- * ext/mplex/gstmplex.h:
- * ext/mplex/gstmplex.hh:
- * ext/mplex/gstmplexibitstream.cc:
- * ext/mplex/gstmplexibitstream.hh:
- * ext/mplex/gstmplexjob.cc:
- * ext/mplex/gstmplexjob.hh:
- * ext/mplex/gstmplexoutputstream.cc:
- * ext/mplex/gstmplexoutputstream.hh:
- We wrap mjpegtools mplex. So I rewrote the plugin. The old plugin
- had issues, didn't do capsnego, supported only a subset of the
- mplex features and required a mplex fork in our local CVS. Plus
- that it worked agaist a very old mplex version. Rewriting was
- faster than updating it.
- * gst-libs/ext/Makefile.am:
- * gst-libs/ext/mplex/INSTRUCT:
- * gst-libs/ext/mplex/Makefile.am:
- * gst-libs/ext/mplex/README:
- * gst-libs/ext/mplex/TODO:
- * gst-libs/ext/mplex/ac3strm_in.cc:
- * gst-libs/ext/mplex/audiostrm.hh:
- * gst-libs/ext/mplex/audiostrm_out.cc:
- * gst-libs/ext/mplex/aunit.hh:
- * gst-libs/ext/mplex/bits.cc:
- * gst-libs/ext/mplex/bits.hh:
- * gst-libs/ext/mplex/buffer.cc:
- * gst-libs/ext/mplex/buffer.hh:
- * gst-libs/ext/mplex/fastintfns.h:
- * gst-libs/ext/mplex/format_codes.h:
- * gst-libs/ext/mplex/inputstrm.cc:
- * gst-libs/ext/mplex/inputstrm.hh:
- * gst-libs/ext/mplex/lpcmstrm_in.cc:
- * gst-libs/ext/mplex/mjpeg_logging.cc:
- * gst-libs/ext/mplex/mjpeg_logging.h:
- * gst-libs/ext/mplex/mjpeg_types.h:
- * gst-libs/ext/mplex/mpastrm_in.cc:
- * gst-libs/ext/mplex/mpegconsts.cc:
- * gst-libs/ext/mplex/mpegconsts.h:
- * gst-libs/ext/mplex/mplexconsts.hh:
- * gst-libs/ext/mplex/multplex.cc:
- * gst-libs/ext/mplex/outputstream.hh:
- * gst-libs/ext/mplex/padstrm.cc:
- * gst-libs/ext/mplex/padstrm.hh:
- * gst-libs/ext/mplex/stillsstream.cc:
- * gst-libs/ext/mplex/stillsstream.hh:
- * gst-libs/ext/mplex/systems.cc:
- * gst-libs/ext/mplex/systems.hh:
- * gst-libs/ext/mplex/vector.cc:
- * gst-libs/ext/mplex/vector.hh:
- * gst-libs/ext/mplex/videostrm.hh:
- * gst-libs/ext/mplex/videostrm_in.cc:
- * gst-libs/ext/mplex/videostrm_out.cc:
- * gst-libs/ext/mplex/yuv4mpeg.cc:
- * gst-libs/ext/mplex/yuv4mpeg.h:
- * gst-libs/ext/mplex/yuv4mpeg_intern.h:
- * gst-libs/ext/mplex/yuv4mpeg_ratio.cc:
- We don't fork mjpegtools' mplex in our CVS anymore.
- * gst/avi/gstavidemux.c: (gst_avi_demux_src_getcaps),
- (gst_avi_demux_add_stream):
- * gst/avi/gstavidemux.h:
- Add getcaps() function for proper caps nego. This makes some
- parts of AVI playback/reading work.
- * sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
- Resize window on new capsnego. This is probably wrong, but
- I'm still committing it because with current capsnego, the
- first successfull capsnego is auto-fixated, therefore rounded
- down to the lowest values in the caps. this results in a 16x16
- XWindow that is not reized when real capsnego finishes.
- Dave, I see more cases of this, do you know a proper solution?
- * tools/gst-launch-ext.in:
- Fix MPEG-4 AAC (Apple iPod/iTunes) file commandline.
-
-2003-12-31 David Schleef <ds@schleef.org>
-
- * gst/tcp/gsttcpsrc.c: (gst_tcpsrc_get):
- * gst/udp/gstudpsrc.c: (gst_udpsrc_get):
- Change gst_pad_proxy_link() to gst_pad_try_set_caps()
-
-2003-12-30 David Schleef <ds@schleef.org>
-
- * ext/ffmpeg/gstffmpegcolorspace.c:
- (gst_ffmpegcsp_caps_remove_format_info), (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_pad_link), (gst_ffmpegcsp_init),
- (gst_ffmpegcsp_chain): Negotiation fixes
- * ext/mad/gstmad.c: (gst_mad_chain): Negotiation fixes
- * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
- (gst_audio_convert_link), (gst_audio_convert_channels):
- * gst/audioscale/gstaudioscale.c: (gst_audioscale_getcaps),
- (gst_audioscale_link), (gst_audioscale_get_buffer),
- (gst_audioscale_chain): Negotiation fixes
- * gst/audioscale/gstaudioscale.h:
- * gst/videofilter/gstvideofilter.c:
- (gst_videofilter_format_get_structure), (gst_videofilter_getcaps),
- (gst_videofilter_link), (gst_videofilter_init),
- (gst_videofilter_set_output_size), (gst_videofilter_setup),
- (gst_videofilter_find_format_by_structure):
- * gst/videofilter/gstvideofilter.h: Negotiation fixes
- * gst/videoscale/gstvideoscale.c: (gst_videoscale_getcaps),
- (gst_videoscale_link):
- * gst/videoscale/videoscale.c: (videoscale_get_structure),
- (videoscale_find_by_structure), (gst_videoscale_setup):
- * gst/videoscale/videoscale.h: Negotiation fixes
- * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
- (gst_ximagesink_fixate), (gst_ximagesink_init): Add a fixate
- function, restrict resizing to a multiple of 4 (hack until
- everyone supports odd sizes correctly).
-
-2003-12-29 Colin Walters <walters@verbum.org>
-
- * ext/esd/esdsink.c (gst_esdsink_link): Fix typo; get depth instead of
- signed.
-
-2003-12-30 Jan Schmidt <thaytan@mad.scientist.com>
-
- * ext/sndfile/gstsf.c: (gst_sf_loop):
- Fix warning about discarding const qualifier
-
-2003-12-27 Jeremy Simon <jesimon@libertysurf.fr>
-
- * gst/cutter/gstcutter.c:
- * gst/videoscale/gstvideoscale.c:
- * gst/volenv/gstvolenv.c:
- * gst-libs/gst/audio/audio.c:
- * gst-libs/gst/video/video.c:
- Fix warnings
+2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ tests/check/generic/states.c: Blacklist cdparanoia element in state test.
+ Original commit message from CVS:
+ * tests/check/generic/states.c: (GST_START_TEST):
+ Blacklist cdparanoia element in state test.
- * ext/alsa/gstalsa.c: (gst_alsa_open_audio):
- Don't send ALSA debugging to stderr.
- * ext/alsa/gstalsa.h:
- Use GST_WARNING instead of g_warning when ALSA functions fail.
+2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
+ Original commit message from CVS:
+ * gst/tcp/gsttcp.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ Add <string.h> includes for memset and FD_ZERO (fixes #323878;
+ patch by: Benjamin Pineau).
- * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
- Free XVAdapterInfo correctly.
+2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
+ Original commit message from CVS:
+ * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
+ (gst_video_rate_chain):
+ Fix timestamping for videorate when the first buffer it sees has a
+ non-zero timestamp. Fix some misleading debug output.
- * ext/mad/gstid3tag.c: (gst_id3_tag_add_src_pad),
- (gst_id3_tag_do_caps_nego), (gst_id3_tag_src_link):
- Make id3tag use correct caps nego.
+2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
-2003-12-27 Amaury Jacquot <sxpert@esitcom.org>
+ gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
+ Original commit message from CVS:
+ * gst/audioresample/gstaudioresample.c:
+ Don't leak all input buffers to audioresample.
- * ext/ivorbis/vorbis.c:
- * ext/ivorbis/vorbisenc.h:
- * ext/ivorbis/vorbisfile.c:
- Modify so that it uses the new caps things
+2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-27 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
+ Original commit message from CVS:
+ * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
+ Don't operate on empty text buffers. Strip newlines and
+ tabs only from the end of the text, but leave them intact
+ in the middle. Fix typo in gtk-doc description.
- * testsuite/spider/spider1.c: (main):
- * testsuite/spider/spider2.c: (main):
- * testsuite/spider/spider3.c: (main):
- Make tests compile again. They probably don't work.
+2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-24 Colin Walters <walters@verbum.org>
+ gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c:
+ * gst/playback/gstplaybin.c: (handoff):
+ Make sure the video frame buffer we return to apps via the
+ "frame" property always has caps set on it. Modify
+ _gst_gvalue_set_object() macro to handle NULL objects
+ gracefully too.
- * sys/oss/gstosssink.c (gst_osssink_sink_fixate): Return NULL if
- we can't fixate the caps anymore.
+2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
+
+ gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
+ Original commit message from CVS:
+ * gst/audiotestsrc/gstaudiotestsrc.c:
+ (gst_audio_test_src_class_init), (gst_audio_test_src_init),
+ (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
+ (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
+ (gst_audio_test_src_create):
+ * gst/audiotestsrc/gstaudiotestsrc.h:
+ Adjust to some recent api changes and add wtays new cool seeking
+ capabillities
+
+2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
+ Original commit message from CVS:
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsadeviceprobe.h:
+ Helper functions to add device probing via the GstPropertyProbe
+ interface to a class.
+ * ext/alsa/gstalsamixer.h:
+ Comment out GST_ALSA_MIXER, it returns a struct that's not
+ used.
+ * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
+ Add some debug info.
+ * ext/alsa/gstalsamixerelement.c:
+ (gst_alsa_mixer_element_interface_supported),
+ (gst_implements_interface_init),
+ (gst_alsa_mixer_element_init_interfaces),
+ (gst_alsa_mixer_element_class_init),
+ (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
+ (gst_alsa_mixer_element_set_property),
+ (gst_alsa_mixer_element_get_property),
+ (gst_alsa_mixer_element_change_state):
+ * ext/alsa/gstalsamixerelement.h:
+ Add 'device' and 'device-name' properties. Add GstPropertyProbe
+ for device handling (gnome-volume-control will need that).
-2003-12-23 David Schleef <ds@schleef.org>
+2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
- * gst/volume/gstvolume.c: (volume_init): Proxy getcaps.
- * sys/oss/gstosssink.c: (gst_osssink_init),
- (gst_osssink_sink_fixate): Add fixate function.
+ * ChangeLog:
+ * ext/Makefile.am:
+ * gst-plugins-base.spec.in:
+ updates to activate cdparanoia plugin
+ Original commit message from CVS:
+ updates to activate cdparanoia plugin
-2003-12-24 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
- * ext/ffmpeg/gstffmpegcodecmap.c:
- * ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_getcaps),
- (gst_ffmpegcsp_srcconnect_func), (gst_ffmpegcsp_sinkconnect),
- (gst_ffmpegcsp_srcconnect), (gst_ffmpegcsp_get_type),
- (gst_ffmpegcsp_base_init), (gst_ffmpegcsp_class_init),
- (gst_ffmpegcsp_init), (gst_ffmpegcsp_chain),
- (gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
- (gst_ffmpegcsp_get_property), (gst_ffmpegcsp_register):
- fix typo in RGB masks, and move back to "old" colorspace
- capsnego code until whoever wrote this new crap has actually
- tested it so that it works.
- And yes, this works, keep it that way please.
+ ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
+ Original commit message from CVS:
+ * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
+ Use the correct function to free list of typefind factories.
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
+ Original commit message from CVS:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ (gst_video_test_src_class_init), (gst_video_test_src_init),
+ (gst_video_test_src_parse_caps), (gst_video_test_src_query),
+ (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
+ (gst_video_test_src_create):
+ * gst/videotestsrc/gstvideotestsrc.h:
+ Implement seeking in videotestsrc.
+ Small cleanups.
+
+2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
+ Original commit message from CVS:
+ * ext/cdparanoia/Makefile.am:
+ * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
+ (gst_paranoia_endian_get_type), (_do_init),
+ (cdparanoia_class_init), (cdparanoia_init),
+ (cdparanoia_set_property), (cdparanoia_get_property),
+ (cdparanoia_do_seek), (cdparanoia_is_seekable),
+ (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
+ (cdparanoia_convert), (cdparanoia_get_query_types),
+ (cdparanoia_query), (cdparanoia_set_index),
+ (cdparanoia_uri_set_uri):
+ * ext/cdparanoia/gstcdparanoia.h:
+ Partially ported cdparanoia now that basesrc can support a
+ plugin like this..
+
+2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
+
+ tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
+ Original commit message from CVS:
+ * tests/examples/seek/scrubby.c: (main):
+ Set higher priority for bus events so they don't get reordered with
+ gtk gui events.
+ * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
+ (flush_toggle_cb), (main):
+ Added checkbox do disable flushing seeks.
+ Disable scrubbing when doing non flushing seeks.
+
+2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
+ Original commit message from CVS:
+ * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
+ (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
+ (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
+ (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
+ Implement some sort of event handling that doesn't rely on
+ g_return_if_fail; make sure we always push the last chunk of an
+ .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
+ state change function; remove some old cruft. Seeking is still
+ rather unlikely to work though.
+ * tools/.cvsignore:
+ Ignore more.
- * ext/divx/gstdivxdec.c: (gst_divxdec_base_init),
- (gst_divxdec_init), (gst_divxdec_negotiate):
- * ext/divx/gstdivxdec.h:
- * ext/divx/gstdivxenc.c: (gst_divxenc_base_init),
- (gst_divxenc_init):
- * ext/faac/gstfaac.c: (gst_faac_base_init), (gst_faac_init),
- (gst_faac_sinkconnect), (gst_faac_srcconnect):
- * ext/mpeg2enc/gstmpeg2enc.cc:
- * ext/mpeg2enc/gstmpeg2encoder.cc:
- * ext/mpeg2enc/gstmpeg2encpicturereader.cc:
- * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_base_init),
- (dxr3audiosink_init), (dxr3audiosink_pcm_sinklink):
- * sys/dxr3/dxr3spusink.c: (dxr3spusink_base_init),
- (dxr3spusink_init):
- * sys/dxr3/dxr3videosink.c: (dxr3videosink_base_init),
- (dxr3videosink_init):
- Fix caps breakage after Dave's caps branch merge.
+2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
-2003-12-23 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
+ Original commit message from CVS:
+ 2005-12-11 Julien MOUTTE <julien@moutte.net>
+ * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
+ Fixed a leak of the current image reference when cleaning up.
+ Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
- * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
- Fix for 24bpp display.
+2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
-2003-12-23 Colin Walters <walters@verbum.org>
+ tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
+ Original commit message from CVS:
+ * tools/Makefile.am:
+ * tools/gst-launch-ext-m.m:
+ Remove gst-launch-ext. It doesn't work, and is no longer
+ particularly useful.
- * ext/gnomevfs/gstgnomevfssink.c: Add ARG_HANDLE property that takes
- a GnomeVFSHandle directly.
+2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
+ Original commit message from CVS:
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
+ don't pass random values to ogmparse convert function.
+ Make seeking possible in the exile1.ogm file.
- * gst/volume/Makefile.am:
- * gst/volume/gstvolume.c: (volume_connect), (volume_parse_caps),
- (volume_base_init), (volume_init):
- Reenable volume element and fix to work with new caps stuff.
- Rhythmbox needs this.
+2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
+ Original commit message from CVS:
+ * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
+ * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
+ Work around refcount problem with g_value_set_object() that occur
+ if the core has been compiled against GLib-2.6 (g_value_set_object()
+ will only g_object_ref() the element, but the caller will
+ gst_object_unref() it and bad things will happen due to the way
+ GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
+ totem for people on FC4 using Thomas's 0.10 RPMs.
- * gst/qtdemux/qtdemux.c: (plugin_init):
- qtdemux requires bytestream
+2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ Time to welcome ogm to 0.10 :)
+ Original commit message from CVS:
+ Time to welcome ogm to 0.10 :)
+ * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
+ (gst_ogg_pad_typefind):
+ Oggdemux can now properly typefind elements with dynamic pads.
+ * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
+ Properly set caps on src pad, and set caps on outgoing buffers.
- * ext/alsa/gstalsa.c: (gst_alsa_get_caps), (gst_alsa_link):
- Fix remaining caps handling errors due to CAPS merge.
+2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-22 Benjamin Otte <in7y118@public.uni-hamburg.de>
+ * ChangeLog:
+ * ext/alsa/gstalsamixer.h:
+ * ext/alsa/gstalsamixerelement.h:
+ * ext/alsa/gstalsamixeroptions.h:
+ * ext/alsa/gstalsamixertrack.h:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasink.h:
+ * ext/alsa/gstalsasrc.c:
+ * ext/alsa/gstalsasrc.h:
+ * ext/cdparanoia/gstcdparanoia.h:
+ * ext/gnomevfs/gstgnomevfsuri.h:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gsttextoverlay.h:
+ * ext/theora/theoradec.c:
+ * ext/theora/theoraenc.c:
+ * ext/vorbis/vorbisdec.h:
+ * ext/vorbis/vorbisenc.c:
+ * ext/vorbis/vorbisenc.h:
+ * ext/vorbis/vorbisparse.h:
+ * gst-libs/gst/audio/gstaudioclock.h:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosink.h:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstaudiosrc.h:
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.h:
+ * gst-libs/gst/audio/gstringbuffer.h:
+ * gst-libs/gst/audio/multichannel.h:
+ * gst-libs/gst/floatcast/floatcast.h:
+ * gst-libs/gst/interfaces/colorbalance.c:
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/colorbalancechannel.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ * gst-libs/gst/interfaces/mixeroptions.h:
+ * gst-libs/gst/interfaces/mixertrack.h:
+ * gst-libs/gst/interfaces/navigation.h:
+ * gst-libs/gst/interfaces/propertyprobe.h:
+ * gst-libs/gst/interfaces/tuner.h:
+ * gst-libs/gst/interfaces/tunerchannel.h:
+ * gst-libs/gst/interfaces/tunernorm.h:
+ * gst-libs/gst/interfaces/xoverlay.h:
+ * gst-libs/gst/netbuffer/gstnetbuffer.h:
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-media.h:
+ * gst-libs/gst/riff/riff-read.h:
+ * gst-libs/gst/rtp/gstbasertpdepayload.h:
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * gst-libs/gst/tag/gsttageditingprivate.h:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ * gst-libs/gst/tag/tag.h:
+ * gst-libs/gst/video/video.h:
+ * gst/adder/gstadder.c:
+ * gst/adder/gstadder.h:
+ * gst/audioconvert/audioconvert.c:
+ * gst/audioconvert/audioconvert.h:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioconvert/gstchannelmix.c:
+ * gst/audioconvert/gstchannelmix.h:
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/resample.h:
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstplaybasebin.h:
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gststreaminfo.h:
+ * gst/tcp/gstfdset.c:
+ * gst/tcp/gstfdset.h:
+ * gst/tcp/gstmultifdsink.c:
+ * gst/tcp/gstmultifdsink.h:
+ * gst/tcp/gsttcp.h:
+ * gst/tcp/gsttcpclientsrc.c:
+ * gst/tcp/gsttcpclientsrc.h:
+ * gst/tcp/gsttcpplugin.h:
+ * gst/tcp/gsttcpserversink.c:
+ * gst/tcp/gsttcpserversrc.c:
+ * gst/typefind/gsttypefindfunctions.c:
+ * gst/videorate/gstvideorate.c:
+ * gst/videotestsrc/gstvideotestsrc.h:
+ * gst/videotestsrc/videotestsrc.h:
+ * sys/v4l/gstv4lcolorbalance.h:
+ * sys/v4l/gstv4ltuner.h:
+ * sys/v4l/gstv4lxoverlay.h:
+ * sys/v4l/v4l_calls.h:
+ * sys/v4l/videodev_mjpeg.h:
+ * tests/check/elements/audioconvert.c:
+ * tests/check/elements/audioresample.c:
+ * tests/check/elements/audiotestsrc.c:
+ * tests/check/elements/videotestsrc.c:
+ * tests/check/elements/volume.c:
+ * tests/examples/seek/scrubby.c:
+ * tests/examples/seek/seek.c:
+ expand tabs
+ Original commit message from CVS:
+ expand tabs
- * ext/faad/gstfaad.c: (gst_faad_base_init), (gst_faad_init),
- (gst_faad_sinkconnect), (gst_faad_srcgetcaps),
- (gst_faad_srcconnect):
- Port to new caps system.
+2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-21 Julien MOUTTE <julien@moutte.net>
+ * docs/libs/tmpl/gstaudio.sgml:
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/libs/tmpl/gstgconf.sgml:
+ * docs/libs/tmpl/gstmixer.sgml:
+ * docs/libs/tmpl/gstringbuffer.sgml:
+ * docs/libs/tmpl/gsttuner.sgml:
+ * docs/libs/tmpl/gstxoverlay.sgml:
+ put back stability level
+ Original commit message from CVS:
+ put back stability level
- * examples/gstplay/player.c: (got_time_tick), (got_stream_length),
- (got_video_size), (main): using g_print instead of g_message.
- * gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup): Fixing EOS
- signal which was not emitted because of "switch" element added to the
- bin but not connected. (Removing from the bin temporarily)
+2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
-2003-12-21 Julien MOUTTE <julien@moutte.net>
+ * configure.ac:
+ back to HEAD
+ Original commit message from CVS:
+ back to HEAD
- * configure.ac: X_DISPLAY_MISSING is set to 1 if AC_PATH_XTRA fails to
- find X development files. I don't understand the previous tests and
- they fail on my debian/ppc unstable. This one works.
- * examples/gstplay/player.c: (main): Set the pipeline to READY before
- exiting.
- * gst-libs/gst/play/gstplay.c: (gst_play_get_length_callback),
- (gst_play_set_video_sink), (gst_play_set_audio_sink),
- (gst_play_set_visualization): Add some safety checks in set_ methods
- and state_change. This was throwing some ugly CRITICAL messages when
- pipeline was getting disposed and casts were failing.
+=== release 0.10.0 ===
-2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
+2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- Improve mpeg2enc detection. This is for distributions that do
- ship mjpegtools, but without mpeg2enc. Also does object check
- for might there ever be ABI incompatibility.
- * ext/mpeg2enc/gstmpeg2enc.cc:
- Add Andrew as second maintainer (he's helping me), and also add
- an error if no caps was set. This happens if I pull before capsnego
- and that's something I should solve sometime else.
- * gst/matroska/matroska-demux.c:
- (gst_matroska_demux_parse_blockgroup):
- Fix time parsing.
- * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
- (gst_matroska_mux_track_header):
- Add caps to templates.
- * gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
- Add mpegversion=1 to prevent confusion with MPEG/AAC.
- * gst/mpegstream/gstmpegdemux.c:
- Remove layer since it causes warnings about unfixed caps.
- * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
- Fix obvious typo (we error out if caps were set, we should of
- course error out if *no* caps were set).
- * sys/oss/gstosselement.c: (gst_osselement_convert):
- Fix format conversion, we confused bits/bytes.
- * sys/oss/gstosselement.h:
- Improve documentation for 'bps'.
- * sys/v4l/TODO:
- Remove stuff about plugins that need removing - this was done
- ages ago.
- * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
- (gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
- * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
- (gst_v4lsrc_src_query):
- * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
- (gst_v4l2src_src_convert), (gst_v4l2src_src_query):
- Add get_query_types(), get_formats() and query() functions.
-
-2003-12-21 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * ChangeLog: moved to gstreamer/docs/random/old/ChangeLog.gst-plugins
- * moved CVS to freedesktop.org
+ * docs/libs/tmpl/gstcolorbalance.sgml:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-decodebin.xml:
+ * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
+ * docs/plugins/inspect/plugin-gnomevfs.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playbin.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-video4linux.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ releasing 0.10.0
+ Original commit message from CVS:
+ releasing 0.10.0
+
projects / platform / upstream / gstreamer.git / blobdiff