=== release 0.10.25 === 2009-10-05 Jan Schmidt <jan.schmidt@sun.com> * configure.ac: releasing 0.10.25, "Standard disclaimers apply" 2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net> * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: Update .po files 2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net> * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: 0.10.24.4 pre-release 2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB 2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/playback/gstplaysink.c: playsink: make the lock recursive for now Fixes #583255 2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/playback/gstplaysink.c: playsink: fix the vis property getter 2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk> * gst-plugins-base.spec.in: Add missing file to spec file 2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/cdda/gstcddabasesrc.c: * tests/check/libs/cddabasesrc.c: cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc 2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org> * 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. 2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net> * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: 0.10.24.3 pre-release 2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com> * gst-libs/gst/tag/gstvorbistag.c: vorbistag: don't ever return NULL in list of strings. 2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com> * gst/playback/gstplaysink.c: playsink: Expose mute,volume,vis-plugin and font-desc properties https://bugzilla.gnome.org/show_bug.cgi?id=594623 2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com> * gst/playback/gstplaysink.c: GstPlaySink: Expose 'reconfigure' as an action signal. 2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com> * gst/playback/gstplaysink.c: GstPlaySink: Expose flags as a gobject property. 2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.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. 2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net> * docs/libs/gst-plugins-base-libs.types: docs: add new gst_stream_volume_get_type to types file This is needs to get Gobject features to show up in the docs. 2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org> * 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. 2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net> * ext/theora/theoraenc.c: theoraenc: Fix a string leak in _getcaps() 2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net> * 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 2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net> * tests/check/elements/audioresample.c: check: Improve audioresample test Make the audioresample test work with CK_FORK=no, and turn a g_print into a GST_INFO. 2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org> * 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. 2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstplaybin2.c: playbin2: Implement GstStreamVolume interface 2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/volume/gstvolume.c: * gst/volume/gstvolume.h: * tests/check/Makefile.am: * tests/check/elements/volume.c: volume: Implement GstStreamVolume interface 2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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 2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/videorate/gstvideorate.c: videorate: Add Since marker for the new skip-to-first property 2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk> * 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. 2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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 2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net> * 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. 2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: playbin(2): Document that the volume property uses a linear scale Fixes bug #571610. 2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: add ugly backward compat hack Check for pulsesink < 0.10.17 because it includes code that is now included in baseaudiosink. Disable that code in baseaudiosink to be compatible with the older version. 2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org> * 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 2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org> * 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 2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org> * 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. 2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/ogg/gstoggdemux.c: oggdemux: Handle kate and cmml as sparse streams too 2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk> * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggdemux.h: oggdemux: Better handling of sparse streams by sending segment updates Fixes bug #397419. 2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net> * gst/playback/gsturidecodebin.c: docs: tell a biit more about uri-decodebin and buffering 2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/audio/gstaudioclock.c: audioclock: add some more debug 2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/theora/theoraenc.c: theoraenc: Print a debug message with supported formats 2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org> * 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. 2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org> * ext/theora/theoraenc.c: theora: Implement support in theoraenc for Y444 and Y42B Fixes bug #594165. 2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org> * ext/theora/theoraenc.c: theora: Refactor the buffer copy code 2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org> * ext/theora/theoraenc.c: theora: Split yuv_buffer creation into its own function 2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org> * ext/theora/theoraenc.c: theora: Split out buffer resize in its own function 2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org> * 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. 2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * tests/examples/seek/seek.c: seek: make stop state configurable Make it easy to experiment with different stop states (NULL and READY) 2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/audio/gstbaseaudiosink.h: baseaudiosink: whitespace fixes 2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/audio/gstringbuffer.c: ringbuffer: add more debug 2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/interfaces/colorbalance.h: * gst-libs/gst/interfaces/mixer.h: whitespace fixes 2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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 2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc 2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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() 2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com> * gst-libs/gst/interfaces/navigation.c: navigation: Fix compiler warning with MSVC Fixes bug #594275. 2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com> * gst-libs/gst/rtp/gstbasertpdepayload.c: basertpdepayload: fix event forwarding 2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com> * gst-libs/gst/rtp/gstrtcpbuffer.c: rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB Fixes #594258 2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.c: * gst/playback/gstplaysink.h: fix whitespace 2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com> * gst-libs/gst/audio/gstbaseaudiosrc.c: baseaudiosrc: improve slave skew resync The old one did the mistake of not actually advancing the ringbuffer, it just adjusted the segbase, introducing the whole lenght of the ringbuffer as an extra delay in the pipeline. Also make sure that the resync can never go back in time, producing the same timestamps that has already been produced, as this can cause severe problems for sinks and other synching mechanisms. Fixes #594256 2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de> * sys/ximage/ximagesink.c: ximagesink: fix small memory leak when setting window title 2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de> * sys/xvimage/xvimagesink.c: xvimagesink: fix small memory leak when setting window title 2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * .gitignore: introspection: Add *.gir and *.typelib to .gitignore 2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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 2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/rtsp/Makefile.am: rtsp: Fix introspection build by ordering sources/headers in dependency order 2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/audio/Makefile.am: audio: Remove debug echo 2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/audio/Makefile.am: audio: Fix build of introspection data by using dependency order for the headers/sources 2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/app/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/cdda/Makefile.am: * gst-libs/gst/fft/Makefile.am: * gst-libs/gst/interfaces/Makefile.am: * gst-libs/gst/netbuffer/Makefile.am: * gst-libs/gst/pbutils/Makefile.am: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtsp/Makefile.am: * gst-libs/gst/sdp/Makefile.am: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/video/Makefile.am: introspection: Strip Gst prefix from all types/functions 2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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 2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/video/Makefile.am: video: Add gobject-introspection support 2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/tag/Makefile.am: tag: Add gobject-introspection support 2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/sdp/Makefile.am: sdp: Add gobject-introspection support 2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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 2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/rtsp/Makefile.am: rtsp: Add gobject-introspection support 2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/rtp/Makefile.am: rtp: Add gobject-introspection support 2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/riff/Makefile.am: riff: Add gobject-introspection support 2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/pbutils/Makefile.am: pbutils: Add gobject-introspection support 2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/netbuffer/Makefile.am: netbuffer: Add gobject-introspection support 2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/interfaces/Makefile.am: interfaces: Add gobject-introspection support 2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/fft/Makefile.am: fft: Add gobject-introspection support 2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/cdda/Makefile.am: cdda: Add gobject-introspection support This is disabled for now until gobject-introspection is fixed 2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/audio/Makefile.am: audio: Add gobject-introspection support 2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * configure.ac: * gst-libs/gst/app/Makefile.am: app: Add gobject-introspection support 2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * common: Automatic update of common submodule From 00a859e to 19fa4f3 2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppay: add some debugging 2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppay: fix frame duration calculations Fix the calculation of the frame duration and rtp timestamps. Add some debugging 2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertppayload.c: rtppay: add some debugging 2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppay: move function around 2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppay: fix sample duration calculation 2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppay: more refactoring Unify the sample/frame buffer handling code by making the functions plugable. 2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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() 2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * common: Update common 2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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. 2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertppayload.h: basertppay: add macro to cast 2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiopayload: code cleanups 2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertpaudiopayload.c: audiortppayload: don't check adapter the adapter is never NULL so we don't need to check it. Use _scale functions to avoid overflows. 2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/subparse/gstsubparse.c: subparse: GstAdapter is not a GstObject and should be freed with g_object_unref 2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * sys/v4l/v4lsrc_calls.c: v4lsrc: fix timestamping for when we do not have a clock yet Should fix #559049. 2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * sys/v4l/v4lsrc_calls.c: v4lsrc: don't log not-yet-initialised integer value 2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * sys/v4l/v4lsrc_calls.c: v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize And reflow code to be more indent friendly. 2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com> * 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. 2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.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. 2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstdecodebin2.c: decodebin2: Post missing plugin messages before any error messages 2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/cdda/gstcddabasesrc.c: cddabasesrc: safely handle the indexes 2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * win32/common/libgstrtsp.def: def: add new rtsp symbols 2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/rtp/gstbasertppayload.h: basertppayload: whitespace fixes. 2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com> * gst/gdp/gstgdppay.c: Bug 593035 - set IN_CAPS for streamheader buffer 2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstinputselector.c: * gst/playback/gststreamselector.c: playbin: The internally linked pad of the selector might be NULL in some cases 2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst-libs/gst/riff/riff-ids.h: * gst-libs/gst/riff/riff-read.c: riff: Add support for AVF files AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF. Fixes bug #593117. 2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * tests/check/elements/audioresample.c: audioresample: Add unit test for checking for timestamp drifts This also checks for perfect timestamping and offsetting. 2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * tests/check/elements/audioresample.c: audioresample: Improve debugging a bit in the unit test 2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com> * configure.ac: * gst/subparse/gstsubparse.c: subparse: Remove dependency on regex.h as it's not used anyway Fixes bug #592544. 2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu> * gst/audioresample/gstaudioresample.c: audioresample: Fix buffer overflow when pushing the drain 2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu> * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: audioresample: Fix timestamp drift Fixes bug #591934. 2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org> * 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. 2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspdefs.c: rtsp: Mark Transport as supporting multiple values. 2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.h: * gst-libs/gst/rtsp/gstrtspdefs.h: * gst-libs/gst/rtsp/gstrtspmessage.h: rtsp: Added missing Since tags. 2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.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 2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.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> * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Made gst_rtsp_watch_queue_data() thread safe. 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.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() 2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com> * 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). 2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspdefs.c: * gst-libs/gst/rtsp/gstrtspdefs.h: rtsp: Do not split headers which should not be split. From RFC 2068 section 4.2: "Multiple message-header fields with the same field-name may be present in a message if and only if the entire field-value for that header field is defined as a comma-separated list [i.e., #(values)]." This means that we should not split other headers which may contain a comma, e.g., Range and Date. 2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com> * 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. 2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Improve parse_line(). Make parse_line() handle keys with multiple values on one line correctly. 2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Normalize lines (remove extra whitespace) before parsing. 2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Improved parsing of messages. Do not abort message parsing as soon as there is an error. Instead parse as much as possible to allow a server to return as meaningful an error as possible. 2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspdefs.c: * gst-libs/gst/rtsp/gstrtspdefs.h: * gst-libs/gst/rtsp/gstrtspmessage.c: * gst-libs/gst/rtsp/gstrtspmessage.h: rtsp: Added support for HTTP messages 2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com> * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspconnection.h: rtsp: Added gst_rtsp_connection_create_from_fd(). API: gst_rtsp_connection_create_from_fd() 2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.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. 2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * 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 2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * docs/libs/gst-plugins-base-libs-sections.txt: docs: fix includes for appsrc/appsink 2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.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). 2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstinputselector.c: * gst/playback/gststreamselector.c: streamselector/inputselector: Use iterate internal links instead of deprecated get internal links 2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com> * 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). 2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com> * 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... 2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gstplaysink.c: playsink: Also send SEEK events directly to a subpicture sink 2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gsturidecodebin.c: uridecodebin: Post a correct error message for unknown types Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN because a plugin is missing and nothing else is wrong. Also make it an error instead of a warning. Really fixes bug #591677. 2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/playback/gsturidecodebin.c: uridecodebin: Post a missing plugin message additional to the error message on unknown types Fixes bug #591677. 2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/playback/gstplaysink.c: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: playbin2: fix error message string Fixes #591577. 2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> * gst-libs/gst/riff/riff-read.c: riff: align API doc of gst_riff_parse_chunk with reality 2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> * gst/playback/gstdecodebin2.c: decodebin2: avoid assertion failure on empty/NULL caps 2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Also detect SVG by the <svg> starting tag Not all SVG images have the DOCTYPE specified. 2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * tests/check/pipelines/streamheader.c: streamheader: Fix caps leak in the vorbisenc unit test 2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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 2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org> * gst-libs/gst/interfaces/navigation.c: navigation: Fix doc blurb typo for gst_navigation_send_key_event 2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/subparse/gstsubparse.c: subparse: Allow . instead of , as millisecond delimiter in srt subtitles Fixes bug #591207. 2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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 2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com> * gst-libs/gst/audio/gstaudioclock.c: * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosrc.c: * gst-libs/gst/audio/gstbaseaudiosrc.c: * gst-libs/gst/audio/gstringbuffer.c: * gst-libs/gst/interfaces/propertyprobe.c: * gst-libs/gst/riff/riff-media.c: * gst-libs/gst/rtp/gstbasertpdepayload.c: * gst-libs/gst/video/gstvideofilter.c: * gst-libs/gst/video/gstvideosink.c: gst-libs: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com> * 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. 2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com> * 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 2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com> * docs/design/draft-va.txt: docs: add draft for generic introduction of video acceleration APIs idea 2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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). 2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * 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. 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org> * ext/theora/gsttheoradec.h: * ext/theora/theoradec.c: theora: Convert theoradec to libtheora 1.0 API 2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextrender.c: textrender: Fix blitting of text over the output buffer and cairo painting 2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextrender.c: textrender: Fix endianness problems (i.e. make it work again on big endian architectures) 2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net> * tests/icles/test-colorkey.c: colorkey-test: fix xsync error 2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com> * gst/ffmpegcolorspace/imgconvert.c: * gst/ffmpegcolorspace/imgconvert_template.h: ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats 2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.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. 2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> * 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. 2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst-libs/gst/audio/gstbaseaudiosrc.c: baseaudiosrc: change default slave method Set the default slave method to the much better skew slaving algortihm. 2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * ext/pango/gsttextoverlay.c: textoverlay: make buffer writable Make the input buffer writable before changing its contents. 2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/typefind/gsttypefindfunctions.c: typefinding: fix postscript typefinder probability Two bytes for a rare format hardly warrants MAXIMUM typefinding probability, POSSIBLE seems more appropriate. 2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextoverlay.c: pango: Send queries from the srcpad directly to the video sinkpad 2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/subparse/gstsubparse.c: subparse: Implement POSITION query 2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/subparse/gstsubparse.c: * gst/subparse/samiparse.c: subparse: Implement SEEKING query 2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com> * 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. 2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextoverlay.c: * ext/pango/gsttextoverlay.h: pango: Add support for xRGB and BGRx formats 2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextoverlay.c: pango: Re-add shading support which was dropped by a previous patch 2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * configure.ac: * ext/pango/gsttextoverlay.c: pango: Check if pangocairo supports vertical rendering and fix properties 2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gsttextrender.c: textrender: Use PROP_X instead of ARG_X consistently 2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: pango: Some minor cleanup 2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * configure.ac: pango: Check for pangocairo instead of pangoft2 2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net> * 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. 2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Add SVG typefinder 2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Add postscript typefinder 2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Use static caps again for MPEG4 typefinding 2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Implement better & more flexible MPEG4 typefinding This detects more MPEG4 streams as MPEG4. Fixes bug #556537. 2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/videoscale/gstvideoscale.c: videoscale: Restrict width/height to 2^15 - 1 Otherwise integer overflows will happen, resulting in segmentation faults. Fixes bug #590243. 2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/ffmpegcolorspace/imgconvert_template.h: ffmpegcolorspace: Fix indention of template header 2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com> * gst-libs/gst/app/gstappsrc.c: appsrc: Clarify documentation about caps and linkage Fixes bug #589095. 2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net> * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Fix typefinding of SDP files Fixes bug #589574. 2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu> * gst/audioresample/gstaudioresample.c: audioresample: Take the output offsets from the input if possible Fixes bug #588915. 2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/videoscale/gstvideoscale.c: videoscale: Make sure to allocate enough memory for the temporary buffer and fix scaling of odd-height interlaced video. 2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * gst/videoscale/gstvideoscale.c: videoscale: Fix interlaced scaling for I420 ...and some other minor mistakes in the previous change. 2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> * 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. 2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net> * configure.ac: back to development -> 0.10.24.1 2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net> * gst-plugins-base.doap: Add 0.10.24 release to the doap file